audio_coding: remove "main" directory

This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}

webrtc/modules/audio_coding/test/ACMTest.h

@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+
+class ACMTest {
+ public:
+  ACMTest() {}
+  virtual ~ACMTest() {}
+  virtual void Perform() = 0;
+};
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_

webrtc/modules/audio_coding/test/APITest.h

@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+
+namespace webrtc {
+
+class Config;
+
+enum APITESTAction {
+  TEST_CHANGE_CODEC_ONLY = 0,
+  DTX_TEST = 1
+};
+
+class APITest : public ACMTest {
+ public:
+  explicit APITest(const Config& config);
+  ~APITest();
+
+  void Perform();
+ private:
+  int16_t SetUp();
+
+  static bool PushAudioThreadA(void* obj);
+  static bool PullAudioThreadA(void* obj);
+  static bool ProcessThreadA(void* obj);
+  static bool APIThreadA(void* obj);
+
+  static bool PushAudioThreadB(void* obj);
+  static bool PullAudioThreadB(void* obj);
+  static bool ProcessThreadB(void* obj);
+  static bool APIThreadB(void* obj);
+
+  void CheckVADStatus(char side);
+
+  // Set Min delay, get delay, playout timestamp
+  void TestDelay(char side);
+
+  // Unregister a codec & register again.
+  void TestRegisteration(char side);
+
+  // Playout Mode, background noise mode.
+  // Receiver Frequency, playout frequency.
+  void TestPlayout(char receiveSide);
+
+  //
+  void TestSendVAD(char side);
+
+  void CurrentCodec(char side);
+
+  void ChangeCodec(char side);
+
+  void Wait(uint32_t waitLengthMs);
+
+  void RunTest(char thread);
+
+  bool PushAudioRunA();
+  bool PullAudioRunA();
+  bool ProcessRunA();
+  bool APIRunA();
+
+  bool PullAudioRunB();
+  bool PushAudioRunB();
+  bool ProcessRunB();
+  bool APIRunB();
+
+  //--- ACMs
+  rtc::scoped_ptr<AudioCodingModule> _acmA;
+  rtc::scoped_ptr<AudioCodingModule> _acmB;
+
+  //--- Channels
+  Channel* _channel_A2B;
+  Channel* _channel_B2A;
+
+  //--- I/O files
+  // A
+  PCMFile _inFileA;
+  PCMFile _outFileA;
+  // B
+  PCMFile _outFileB;
+  PCMFile _inFileB;
+
+  //--- I/O params
+  // A
+  int32_t _outFreqHzA;
+  // B
+  int32_t _outFreqHzB;
+
+  // Should we write to file.
+  // we might skip writing to file if we
+  // run the test for a long time.
+  bool _writeToFile;
+  //--- Events
+  // A
+  EventTimerWrapper* _pullEventA;      // pulling data from ACM
+  EventTimerWrapper* _pushEventA;      // pushing data to ACM
+  EventTimerWrapper* _processEventA;   // process
+  EventWrapper* _apiEventA;       // API calls
+  // B
+  EventTimerWrapper* _pullEventB;      // pulling data from ACM
+  EventTimerWrapper* _pushEventB;      // pushing data to ACM
+  EventTimerWrapper* _processEventB;   // process
+  EventWrapper* _apiEventB;       // API calls
+
+  // keep track of the codec in either side.
+  uint8_t _codecCntrA;
+  uint8_t _codecCntrB;
+
+  // Is set to true if there is no encoder in either side
+  bool _thereIsEncoderA;
+  bool _thereIsEncoderB;
+  bool _thereIsDecoderA;
+  bool _thereIsDecoderB;
+
+  bool _sendVADA;
+  bool _sendDTXA;
+  ACMVADMode _sendVADModeA;
+
+  bool _sendVADB;
+  bool _sendDTXB;
+  ACMVADMode _sendVADModeB;
+
+  int32_t _minDelayA;
+  int32_t _minDelayB;
+  bool _payloadUsed[32];
+
+  bool _verbose;
+
+  int _dotPositionA;
+  int _dotMoveDirectionA;
+  int _dotPositionB;
+  int _dotMoveDirectionB;
+
+  char _movingDot[41];
+
+  VADCallback* _vadCallbackA;
+  VADCallback* _vadCallbackB;
+  RWLockWrapper& _apiTestRWLock;
+  bool _randomTest;
+  int _testNumA;
+  int _testNumB;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_

webrtc/modules/audio_coding/test/Channel.cc

@@ -0,0 +1,424 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/Channel.h"
+
+#include <assert.h>
+#include <iostream>
+
+#include "webrtc/base/format_macros.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+int32_t Channel::SendData(FrameType frameType,
+                          uint8_t payloadType,
+                          uint32_t timeStamp,
+                          const uint8_t* payloadData,
+                          size_t payloadSize,
+                          const RTPFragmentationHeader* fragmentation) {
+  WebRtcRTPHeader rtpInfo;
+  int32_t status;
+  size_t payloadDataSize = payloadSize;
+
+  rtpInfo.header.markerBit = false;
+  rtpInfo.header.ssrc = 0;
+  rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
+      _seqNo++ : static_cast<uint16_t>(external_sequence_number_);
+  rtpInfo.header.payloadType = payloadType;
+  rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
+      static_cast<uint32_t>(external_send_timestamp_);
+
+  if (frameType == kAudioFrameCN) {
+    rtpInfo.type.Audio.isCNG = true;
+  } else {
+    rtpInfo.type.Audio.isCNG = false;
+  }
+  if (frameType == kEmptyFrame) {
+    // When frame is empty, we should not transmit it. The frame size of the
+    // next non-empty frame will be based on the previous frame size.
+    _useLastFrameSize = _lastFrameSizeSample > 0;
+    return 0;
+  }
+
+  rtpInfo.type.Audio.channel = 1;
+  // Treat fragmentation separately
+  if (fragmentation != NULL) {
+    // If silence for too long, send only new data.
+    if ((fragmentation->fragmentationVectorSize == 2) &&
+        (fragmentation->fragmentationTimeDiff[1] <= 0x3fff)) {
+      // only 0x80 if we have multiple blocks
+      _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
+      size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
+          fragmentation->fragmentationLength[1];
+      _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
+      _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
+      _payloadData[3] = uint8_t(REDheader & 0x000000FF);
+
+      _payloadData[4] = fragmentation->fragmentationPlType[0];
+      // copy the RED data
+      memcpy(_payloadData + 5,
+             payloadData + fragmentation->fragmentationOffset[1],
+             fragmentation->fragmentationLength[1]);
+      // copy the normal data
+      memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
+             payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
+      payloadDataSize += 5;
+    } else {
+      // single block (newest one)
+      memcpy(_payloadData, payloadData + fragmentation->fragmentationOffset[0],
+             fragmentation->fragmentationLength[0]);
+      payloadDataSize = fragmentation->fragmentationLength[0];
+      rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
+    }
+  } else {
+    memcpy(_payloadData, payloadData, payloadDataSize);
+    if (_isStereo) {
+      if (_leftChannel) {
+        memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
+        _leftChannel = false;
+        rtpInfo.type.Audio.channel = 1;
+      } else {
+        memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
+        _leftChannel = true;
+        rtpInfo.type.Audio.channel = 2;
+      }
+    }
+  }
+
+  _channelCritSect->Enter();
+  if (_saveBitStream) {
+    //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
+  }
+
+  if (!_isStereo) {
+    CalcStatistics(rtpInfo, payloadSize);
+  }
+  _useLastFrameSize = false;
+  _lastInTimestamp = timeStamp;
+  _totalBytes += payloadDataSize;
+  _channelCritSect->Leave();
+
+  if (_useFECTestWithPacketLoss) {
+    _packetLoss += 1;
+    if (_packetLoss == 3) {
+      _packetLoss = 0;
+      return 0;
+    }
+  }
+
+  if (num_packets_to_drop_ > 0) {
+    num_packets_to_drop_--;
+    return 0;
+  }
+
+  status = _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtpInfo);
+
+  return status;
+}
+
+// TODO(turajs): rewite this method.
+void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
+  int n;
+  if ((rtpInfo.header.payloadType != _lastPayloadType)
+      && (_lastPayloadType != -1)) {
+    // payload-type is changed.
+    // we have to terminate the calculations on the previous payload type
+    // we ignore the last packet in that payload type just to make things
+    // easier.
+    for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+      if (_lastPayloadType == _payloadStats[n].payloadType) {
+        _payloadStats[n].newPacket = true;
+        break;
+      }
+    }
+  }
+  _lastPayloadType = rtpInfo.header.payloadType;
+
+  bool newPayload = true;
+  ACMTestPayloadStats* currentPayloadStr = NULL;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) {
+      newPayload = false;
+      currentPayloadStr = &_payloadStats[n];
+      break;
+    }
+  }
+
+  if (!newPayload) {
+    if (!currentPayloadStr->newPacket) {
+      if (!_useLastFrameSize) {
+        _lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
+            (uint32_t) currentPayloadStr->lastTimestamp);
+      }
+      assert(_lastFrameSizeSample > 0);
+      int k = 0;
+      for (; k < MAX_NUM_FRAMESIZES; ++k) {
+        if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
+            _lastFrameSizeSample) ||
+            (currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
+          break;
+        }
+      }
+      if (k == MAX_NUM_FRAMESIZES) {
+        // New frame size found but no space to count statistics on it. Skip it.
+        printf("No memory to store statistics for payload %d : frame size %d\n",
+               _lastPayloadType, _lastFrameSizeSample);
+        return;
+      }
+      ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
+          ->frameSizeStats[k]);
+      currentFrameSizeStats->frameSizeSample = (int16_t) _lastFrameSizeSample;
+
+      // increment the number of encoded samples.
+      currentFrameSizeStats->totalEncodedSamples += _lastFrameSizeSample;
+      // increment the number of recveived packets
+      currentFrameSizeStats->numPackets++;
+      // increment the total number of bytes (this is based on
+      // the previous payload we don't know the frame-size of
+      // the current payload.
+      currentFrameSizeStats->totalPayloadLenByte += currentPayloadStr
+          ->lastPayloadLenByte;
+      // store the maximum payload-size (this is based on
+      // the previous payload we don't know the frame-size of
+      // the current payload.
+      if (currentFrameSizeStats->maxPayloadLen
+          < currentPayloadStr->lastPayloadLenByte) {
+        currentFrameSizeStats->maxPayloadLen = currentPayloadStr
+            ->lastPayloadLenByte;
+      }
+      // store the current values for the next time
+      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+    } else {
+      currentPayloadStr->newPacket = false;
+      currentPayloadStr->lastPayloadLenByte = payloadSize;
+      currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
+      currentPayloadStr->payloadType = rtpInfo.header.payloadType;
+      memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
+             sizeof(ACMTestFrameSizeStats));
+    }
+  } else {
+    n = 0;
+    while (_payloadStats[n].payloadType != -1) {
+      n++;
+    }
+    // first packet
+    _payloadStats[n].newPacket = false;
+    _payloadStats[n].lastPayloadLenByte = payloadSize;
+    _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
+    _payloadStats[n].payloadType = rtpInfo.header.payloadType;
+    memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
+           sizeof(ACMTestFrameSizeStats));
+  }
+}
+
+Channel::Channel(int16_t chID)
+    : _receiverACM(NULL),
+      _seqNo(0),
+      _channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
+      _bitStreamFile(NULL),
+      _saveBitStream(false),
+      _lastPayloadType(-1),
+      _isStereo(false),
+      _leftChannel(true),
+      _lastInTimestamp(0),
+      _useLastFrameSize(false),
+      _lastFrameSizeSample(0),
+      _packetLoss(0),
+      _useFECTestWithPacketLoss(false),
+      _beginTime(TickTime::MillisecondTimestamp()),
+      _totalBytes(0),
+      external_send_timestamp_(-1),
+      external_sequence_number_(-1),
+      num_packets_to_drop_(0) {
+  int n;
+  int k;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    _payloadStats[n].payloadType = -1;
+    _payloadStats[n].newPacket = true;
+    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+      _payloadStats[n].frameSizeStats[k].numPackets = 0;
+      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+    }
+  }
+  if (chID >= 0) {
+    _saveBitStream = true;
+    char bitStreamFileName[500];
+    sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
+    _bitStreamFile = fopen(bitStreamFileName, "wb");
+  } else {
+    _saveBitStream = false;
+  }
+}
+
+Channel::~Channel() {
+  delete _channelCritSect;
+}
+
+void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
+  _receiverACM = acm;
+  return;
+}
+
+void Channel::ResetStats() {
+  int n;
+  int k;
+  _channelCritSect->Enter();
+  _lastPayloadType = -1;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    _payloadStats[n].payloadType = -1;
+    _payloadStats[n].newPacket = true;
+    for (k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+      _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
+      _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
+      _payloadStats[n].frameSizeStats[k].numPackets = 0;
+      _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
+      _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
+    }
+  }
+  _beginTime = TickTime::MillisecondTimestamp();
+  _totalBytes = 0;
+  _channelCritSect->Leave();
+}
+
+int16_t Channel::Stats(CodecInst& codecInst,
+                       ACMTestPayloadStats& payloadStats) {
+  _channelCritSect->Enter();
+  int n;
+  payloadStats.payloadType = -1;
+  for (n = 0; n < MAX_NUM_PAYLOADS; n++) {
+    if (_payloadStats[n].payloadType == codecInst.pltype) {
+      memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
+      break;
+    }
+  }
+  if (payloadStats.payloadType == -1) {
+    _channelCritSect->Leave();
+    return -1;
+  }
+  for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+    if (payloadStats.frameSizeStats[n].frameSizeSample == 0) {
+      _channelCritSect->Leave();
+      return 0;
+    }
+    payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
+        .frameSizeStats[n].totalEncodedSamples / (double) codecInst.plfreq;
+
+    payloadStats.frameSizeStats[n].rateBitPerSec =
+        payloadStats.frameSizeStats[n].totalPayloadLenByte * 8
+            / payloadStats.frameSizeStats[n].usageLenSec;
+
+  }
+  _channelCritSect->Leave();
+  return 0;
+}
+
+void Channel::Stats(uint32_t* numPackets) {
+  _channelCritSect->Enter();
+  int k;
+  int n;
+  memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
+  for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
+    if (_payloadStats[k].payloadType == -1) {
+      break;
+    }
+    numPackets[k] = 0;
+    for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+      if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
+        break;
+      }
+      numPackets[k] += _payloadStats[k].frameSizeStats[n].numPackets;
+    }
+  }
+  _channelCritSect->Leave();
+}
+
+void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
+  _channelCritSect->Enter();
+
+  int k;
+  int n;
+  memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(uint32_t));
+  for (k = 0; k < MAX_NUM_PAYLOADS; k++) {
+    if (_payloadStats[k].payloadType == -1) {
+      break;
+    }
+    payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
+    payloadLenByte[k] = 0;
+    for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
+      if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
+        break;
+      }
+      payloadLenByte[k] += (uint16_t) _payloadStats[k].frameSizeStats[n]
+          .totalPayloadLenByte;
+    }
+  }
+
+  _channelCritSect->Leave();
+}
+
+void Channel::PrintStats(CodecInst& codecInst) {
+  ACMTestPayloadStats payloadStats;
+  Stats(codecInst, payloadStats);
+  printf("%s %d kHz\n", codecInst.plname, codecInst.plfreq / 1000);
+  printf("=====================================================\n");
+  if (payloadStats.payloadType == -1) {
+    printf("No Packets are sent with payload-type %d (%s)\n\n",
+           codecInst.pltype, codecInst.plname);
+    return;
+  }
+  for (int k = 0; k < MAX_NUM_FRAMESIZES; k++) {
+    if (payloadStats.frameSizeStats[k].frameSizeSample == 0) {
+      break;
+    }
+    printf("Frame-size.................... %d samples\n",
+           payloadStats.frameSizeStats[k].frameSizeSample);
+    printf("Average Rate.................. %.0f bits/sec\n",
+           payloadStats.frameSizeStats[k].rateBitPerSec);
+    printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
+           payloadStats.frameSizeStats[k].maxPayloadLen);
+    printf(
+        "Maximum Instantaneous Rate.... %.0f bits/sec\n",
+        ((double) payloadStats.frameSizeStats[k].maxPayloadLen * 8.0
+            * (double) codecInst.plfreq)
+            / (double) payloadStats.frameSizeStats[k].frameSizeSample);
+    printf("Number of Packets............. %u\n",
+           (unsigned int) payloadStats.frameSizeStats[k].numPackets);
+    printf("Duration...................... %0.3f sec\n\n",
+           payloadStats.frameSizeStats[k].usageLenSec);
+
+  }
+
+}
+
+uint32_t Channel::LastInTimestamp() {
+  uint32_t timestamp;
+  _channelCritSect->Enter();
+  timestamp = _lastInTimestamp;
+  _channelCritSect->Leave();
+  return timestamp;
+}
+
+double Channel::BitRate() {
+  double rate;
+  uint64_t currTime = TickTime::MillisecondTimestamp();
+  _channelCritSect->Enter();
+  rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
+  _channelCritSect->Leave();
+  return rate;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/Channel.h

@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+
+#include <stdio.h>
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class CriticalSectionWrapper;
+
+#define MAX_NUM_PAYLOADS   50
+#define MAX_NUM_FRAMESIZES  6
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestFrameSizeStats {
+  uint16_t frameSizeSample;
+  size_t maxPayloadLen;
+  uint32_t numPackets;
+  uint64_t totalPayloadLenByte;
+  uint64_t totalEncodedSamples;
+  double rateBitPerSec;
+  double usageLenSec;
+};
+
+// TODO(turajs): Write constructor for this structure.
+struct ACMTestPayloadStats {
+  bool newPacket;
+  int16_t payloadType;
+  size_t lastPayloadLenByte;
+  uint32_t lastTimestamp;
+  ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
+};
+
+class Channel : public AudioPacketizationCallback {
+ public:
+
+  Channel(int16_t chID = -1);
+  ~Channel();
+
+  int32_t SendData(FrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  void RegisterReceiverACM(AudioCodingModule *acm);
+
+  void ResetStats();
+
+  int16_t Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats);
+
+  void Stats(uint32_t* numPackets);
+
+  void Stats(uint8_t* payloadType, uint32_t* payloadLenByte);
+
+  void PrintStats(CodecInst& codecInst);
+
+  void SetIsStereo(bool isStereo) {
+    _isStereo = isStereo;
+  }
+
+  uint32_t LastInTimestamp();
+
+  void SetFECTestWithPacketLoss(bool usePacketLoss) {
+    _useFECTestWithPacketLoss = usePacketLoss;
+  }
+
+  double BitRate();
+
+  void set_send_timestamp(uint32_t new_send_ts) {
+    external_send_timestamp_ = new_send_ts;
+  }
+
+  void set_sequence_number(uint16_t new_sequence_number) {
+    external_sequence_number_ = new_sequence_number;
+  }
+
+  void set_num_packets_to_drop(int new_num_packets_to_drop) {
+    num_packets_to_drop_ = new_num_packets_to_drop;
+  }
+
+ private:
+  void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
+
+  AudioCodingModule* _receiverACM;
+  uint16_t _seqNo;
+  // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
+  uint8_t _payloadData[60 * 32 * 2 * 2];
+
+  CriticalSectionWrapper* _channelCritSect;
+  FILE* _bitStreamFile;
+  bool _saveBitStream;
+  int16_t _lastPayloadType;
+  ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
+  bool _isStereo;
+  WebRtcRTPHeader _rtpInfo;
+  bool _leftChannel;
+  uint32_t _lastInTimestamp;
+  bool _useLastFrameSize;
+  uint32_t _lastFrameSizeSample;
+  // FEC Test variables
+  int16_t _packetLoss;
+  bool _useFECTestWithPacketLoss;
+  uint64_t _beginTime;
+  uint64_t _totalBytes;
+
+  // External timing info, defaulted to -1. Only used if they are
+  // non-negative.
+  int64_t external_send_timestamp_;
+  int32_t external_sequence_number_;
+  int num_packets_to_drop_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_

webrtc/modules/audio_coding/test/EncodeDecodeTest.cc

@@ -0,0 +1,351 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+
+#include <sstream>
+#include <stdio.h>
+#include <stdlib.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+TestPacketization::TestPacketization(RTPStream *rtpStream, uint16_t frequency)
+    : _rtpStream(rtpStream),
+      _frequency(frequency),
+      _seqNo(0) {
+}
+
+TestPacketization::~TestPacketization() {
+}
+
+int32_t TestPacketization::SendData(
+    const FrameType /* frameType */, const uint8_t payloadType,
+    const uint32_t timeStamp, const uint8_t* payloadData,
+    const size_t payloadSize,
+    const RTPFragmentationHeader* /* fragmentation */) {
+  _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
+                    _frequency);
+  return 1;
+}
+
+Sender::Sender()
+    : _acm(NULL),
+      _pcmFile(),
+      _audioFrame(),
+      _packetization(NULL) {
+}
+
+void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+                   std::string in_file_name, int sample_rate, int channels) {
+  struct CodecInst sendCodec;
+  int noOfCodecs = acm->NumberOfCodecs();
+  int codecNo;
+
+  // Open input file
+  const std::string file_name = webrtc::test::ResourcePath(in_file_name, "pcm");
+  _pcmFile.Open(file_name, sample_rate, "rb");
+  if (channels == 2) {
+    _pcmFile.ReadStereo(true);
+  }
+
+  // Set the codec for the current test.
+  if ((testMode == 0) || (testMode == 1)) {
+    // Set the codec id.
+    codecNo = codeId;
+  } else {
+    // Choose codec on command line.
+    printf("List of supported codec.\n");
+    for (int n = 0; n < noOfCodecs; n++) {
+      EXPECT_EQ(0, acm->Codec(n, &sendCodec));
+      printf("%d %s\n", n, sendCodec.plname);
+    }
+    printf("Choose your codec:");
+    ASSERT_GT(scanf("%d", &codecNo), 0);
+  }
+
+  EXPECT_EQ(0, acm->Codec(codecNo, &sendCodec));
+
+  sendCodec.channels = channels;
+
+  EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+  _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
+  EXPECT_EQ(0, acm->RegisterTransportCallback(_packetization));
+
+  _acm = acm;
+}
+
+void Sender::Teardown() {
+  _pcmFile.Close();
+  delete _packetization;
+}
+
+bool Sender::Add10MsData() {
+  if (!_pcmFile.EndOfFile()) {
+    EXPECT_GT(_pcmFile.Read10MsData(_audioFrame), 0);
+    int32_t ok = _acm->Add10MsData(_audioFrame);
+    EXPECT_GE(ok, 0);
+    return ok >= 0 ? true : false;
+  }
+  return false;
+}
+
+void Sender::Run() {
+  while (true) {
+    if (!Add10MsData()) {
+      break;
+    }
+  }
+}
+
+Receiver::Receiver()
+    : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
+      _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
+}
+
+void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+                     std::string out_file_name, int channels) {
+  struct CodecInst recvCodec = CodecInst();
+  int noOfCodecs;
+  EXPECT_EQ(0, acm->InitializeReceiver());
+
+  noOfCodecs = acm->NumberOfCodecs();
+  for (int i = 0; i < noOfCodecs; i++) {
+    EXPECT_EQ(0, acm->Codec(i, &recvCodec));
+    if (recvCodec.channels == channels)
+      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+    // Forces mono/stereo for Opus.
+    if (!strcmp(recvCodec.plname, "opus")) {
+      recvCodec.channels = channels;
+      EXPECT_EQ(0, acm->RegisterReceiveCodec(recvCodec));
+    }
+  }
+
+  int playSampFreq;
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath() << out_file_name
+      << static_cast<int>(codeId) << ".pcm";
+  file_name = file_stream.str();
+  _rtpStream = rtpStream;
+
+  if (testMode == 1) {
+    playSampFreq = recvCodec.plfreq;
+    _pcmFile.Open(file_name, recvCodec.plfreq, "wb+");
+  } else if (testMode == 0) {
+    playSampFreq = 32000;
+    _pcmFile.Open(file_name, 32000, "wb+");
+  } else {
+    printf("\nValid output frequencies:\n");
+    printf("8000\n16000\n32000\n-1,");
+    printf("which means output frequency equal to received signal frequency");
+    printf("\n\nChoose output sampling frequency: ");
+    ASSERT_GT(scanf("%d", &playSampFreq), 0);
+    file_name = webrtc::test::OutputPath() + out_file_name + ".pcm";
+    _pcmFile.Open(file_name, playSampFreq, "wb+");
+  }
+
+  _realPayloadSizeBytes = 0;
+  _playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
+  _frequency = playSampFreq;
+  _acm = acm;
+  _firstTime = true;
+}
+
+void Receiver::Teardown() {
+  delete[] _playoutBuffer;
+  _pcmFile.Close();
+  if (testMode > 1) {
+    Trace::ReturnTrace();
+  }
+}
+
+bool Receiver::IncomingPacket() {
+  if (!_rtpStream->EndOfFile()) {
+    if (_firstTime) {
+      _firstTime = false;
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                               _payloadSizeBytes, &_nextTime);
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          _firstTime = true;
+          return true;
+        } else {
+          return false;
+        }
+      }
+    }
+
+    EXPECT_EQ(0, _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes,
+                                      _rtpInfo));
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                             _payloadSizeBytes, &_nextTime);
+    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+      _firstTime = true;
+    }
+  }
+  return true;
+}
+
+bool Receiver::PlayoutData() {
+  AudioFrame audioFrame;
+
+  int32_t ok =_acm->PlayoutData10Ms(_frequency, &audioFrame);
+  EXPECT_EQ(0, ok);
+  if (ok < 0){
+    return false;
+  }
+  if (_playoutLengthSmpls == 0) {
+    return false;
+  }
+  _pcmFile.Write10MsData(audioFrame.data_,
+      audioFrame.samples_per_channel_ * audioFrame.num_channels_);
+  return true;
+}
+
+void Receiver::Run() {
+  uint8_t counter500Ms = 50;
+  uint32_t clock = 0;
+
+  while (counter500Ms > 0) {
+    if (clock == 0 || clock >= _nextTime) {
+      EXPECT_TRUE(IncomingPacket());
+      if (clock == 0) {
+        clock = _nextTime;
+      }
+    }
+    if ((clock % 10) == 0) {
+      if (!PlayoutData()) {
+        clock++;
+        continue;
+      }
+    }
+    if (_rtpStream->EndOfFile()) {
+      counter500Ms--;
+    }
+    clock++;
+  }
+}
+
+EncodeDecodeTest::EncodeDecodeTest() {
+  _testMode = 2;
+  Trace::CreateTrace();
+  Trace::SetTraceFile(
+      (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
+}
+
+EncodeDecodeTest::EncodeDecodeTest(int testMode) {
+  //testMode == 0 for autotest
+  //testMode == 1 for testing all codecs/parameters
+  //testMode > 1 for specific user-input test (as it was used before)
+  _testMode = testMode;
+  if (_testMode != 0) {
+    Trace::CreateTrace();
+    Trace::SetTraceFile(
+        (webrtc::test::OutputPath() + "acm_encdec_trace.txt").c_str());
+  }
+}
+
+void EncodeDecodeTest::Perform() {
+  int numCodecs = 1;
+  int codePars[3];  // Frequency, packet size, rate.
+  int numPars[52];  // Number of codec parameters sets (freq, pacsize, rate)
+                    // to test, for a given codec.
+
+  codePars[0] = 0;
+  codePars[1] = 0;
+  codePars[2] = 0;
+
+  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+  struct CodecInst sendCodecTmp;
+  numCodecs = acm->NumberOfCodecs();
+
+  if (_testMode != 2) {
+    for (int n = 0; n < numCodecs; n++) {
+      EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
+      if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
+        numPars[n] = 0;
+      } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
+        numPars[n] = 0;
+      } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
+        numPars[n] = 0;
+      } else if (sendCodecTmp.channels == 2) {
+        numPars[n] = 0;
+      } else {
+        numPars[n] = 1;
+      }
+    }
+  } else {
+    numCodecs = 1;
+    numPars[0] = 1;
+  }
+
+  _receiver.testMode = _testMode;
+
+  // Loop over all mono codecs:
+  for (int codeId = 0; codeId < numCodecs; codeId++) {
+    // Only encode using real mono encoders, not telephone-event and cng.
+    for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
+      // Encode all data to file.
+      std::string fileName = EncodeToFile(1, codeId, codePars, _testMode);
+
+      RTPFile rtpFile;
+      rtpFile.Open(fileName.c_str(), "rb");
+
+      _receiver.codeId = codeId;
+
+      rtpFile.ReadHeader();
+      _receiver.Setup(acm.get(), &rtpFile, "encodeDecode_out", 1);
+      _receiver.Run();
+      _receiver.Teardown();
+      rtpFile.Close();
+    }
+  }
+
+  // End tracing.
+  if (_testMode == 1) {
+    Trace::ReturnTrace();
+  }
+}
+
+std::string EncodeDecodeTest::EncodeToFile(int fileType,
+                                           int codeId,
+                                           int* codePars,
+                                           int testMode) {
+  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(1));
+  RTPFile rtpFile;
+  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                    "encode_decode_rtp");
+  rtpFile.Open(fileName.c_str(), "wb+");
+  rtpFile.WriteHeader();
+
+  // Store for auto_test and logging.
+  _sender.testMode = testMode;
+  _sender.codeId = codeId;
+
+  _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
+  if (acm->SendCodec()) {
+    _sender.Run();
+  }
+  _sender.Teardown();
+  rtpFile.Close();
+
+  return fileName;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/EncodeDecodeTest.h

@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+
+#include <stdio.h>
+#include <string.h>
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/RTPFile.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+#define MAX_INCOMING_PAYLOAD 8096
+
+// TestPacketization callback which writes the encoded payloads to file
+class TestPacketization : public AudioPacketizationCallback {
+ public:
+  TestPacketization(RTPStream *rtpStream, uint16_t frequency);
+  ~TestPacketization();
+  int32_t SendData(const FrameType frameType,
+                   const uint8_t payloadType,
+                   const uint32_t timeStamp,
+                   const uint8_t* payloadData,
+                   const size_t payloadSize,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+ private:
+  static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
+                            int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
+  RTPStream* _rtpStream;
+  int32_t _frequency;
+  int16_t _seqNo;
+};
+
+class Sender {
+ public:
+  Sender();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string in_file_name, int sample_rate, int channels);
+  void Teardown();
+  void Run();
+  bool Add10MsData();
+
+  //for auto_test and logging
+  uint8_t testMode;
+  uint8_t codeId;
+
+ protected:
+  AudioCodingModule* _acm;
+
+ private:
+  PCMFile _pcmFile;
+  AudioFrame _audioFrame;
+  TestPacketization* _packetization;
+};
+
+class Receiver {
+ public:
+  Receiver();
+  virtual ~Receiver() {};
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string out_file_name, int channels);
+  void Teardown();
+  void Run();
+  virtual bool IncomingPacket();
+  bool PlayoutData();
+
+  //for auto_test and logging
+  uint8_t codeId;
+  uint8_t testMode;
+
+ private:
+  PCMFile _pcmFile;
+  int16_t* _playoutBuffer;
+  uint16_t _playoutLengthSmpls;
+  int32_t _frequency;
+  bool _firstTime;
+
+ protected:
+  AudioCodingModule* _acm;
+  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
+  RTPStream* _rtpStream;
+  WebRtcRTPHeader _rtpInfo;
+  size_t _realPayloadSizeBytes;
+  size_t _payloadSizeBytes;
+  uint32_t _nextTime;
+};
+
+class EncodeDecodeTest : public ACMTest {
+ public:
+  EncodeDecodeTest();
+  explicit EncodeDecodeTest(int testMode);
+  void Perform() override;
+
+  uint16_t _playoutFreq;
+  uint8_t _testMode;
+
+ private:
+  std::string EncodeToFile(int fileType,
+                           int codeId,
+                           int* codePars,
+                           int testMode);
+
+ protected:
+  Sender _sender;
+  Receiver _receiver;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_

webrtc/modules/audio_coding/test/PCMFile.cc

@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "PCMFile.h"
+
+#include <ctype.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+PCMFile::PCMFile()
+    : pcm_file_(NULL),
+      samples_10ms_(160),
+      frequency_(16000),
+      end_of_file_(false),
+      auto_rewind_(false),
+      rewinded_(false),
+      read_stereo_(false),
+      save_stereo_(false) {
+  timestamp_ = (((uint32_t) rand() & 0x0000FFFF) << 16) |
+      ((uint32_t) rand() & 0x0000FFFF);
+}
+
+PCMFile::PCMFile(uint32_t timestamp)
+    : pcm_file_(NULL),
+      samples_10ms_(160),
+      frequency_(16000),
+      end_of_file_(false),
+      auto_rewind_(false),
+      rewinded_(false),
+      read_stereo_(false),
+      save_stereo_(false) {
+  timestamp_ = timestamp;
+}
+
+int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
+                            uint16_t* frequency_hz) {
+  char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
+
+  EXPECT_TRUE(fgets(tmp_name, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
+  tmp_name[MAX_FILE_NAME_LENGTH_BYTE - 1] = '\0';
+  int16_t n = 0;
+
+  // Removing trailing spaces.
+  while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (tmp_name[n] != 0)
+      && (n < MAX_FILE_NAME_LENGTH_BYTE)) {
+    n++;
+  }
+  if (n > 0) {
+    memmove(tmp_name, &tmp_name[n], MAX_FILE_NAME_LENGTH_BYTE - n);
+  }
+
+  // Removing trailing spaces.
+  n = (int16_t)(strlen(tmp_name) - 1);
+  if (n >= 0) {
+    while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (n >= 0)) {
+      n--;
+    }
+  }
+  if (n >= 0) {
+    tmp_name[n + 1] = '\0';
+  }
+
+  int16_t len = (int16_t) strlen(tmp_name);
+  if (len > max_len) {
+    return -1;
+  }
+  if (len > 0) {
+    std::string tmp_string(tmp_name, len + 1);
+    *file_name = tmp_string;
+  }
+  printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
+         *frequency_hz);
+  EXPECT_TRUE(fgets(tmp_name, 10, stdin) != NULL);
+  uint16_t tmp_frequency = (uint16_t) atoi(tmp_name);
+  if (tmp_frequency > 0) {
+    *frequency_hz = tmp_frequency;
+  }
+  return 0;
+}
+
+void PCMFile::Open(const std::string& file_name, uint16_t frequency,
+                   const char* mode, bool auto_rewind) {
+  if ((pcm_file_ = fopen(file_name.c_str(), mode)) == NULL) {
+    printf("Cannot open file %s.\n", file_name.c_str());
+    ADD_FAILURE() << "Unable to read file";
+  }
+  frequency_ = frequency;
+  samples_10ms_ = (uint16_t)(frequency_ / 100);
+  auto_rewind_ = auto_rewind;
+  end_of_file_ = false;
+  rewinded_ = false;
+}
+
+int32_t PCMFile::SamplingFrequency() const {
+  return frequency_;
+}
+
+uint16_t PCMFile::PayloadLength10Ms() const {
+  return samples_10ms_;
+}
+
+int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
+  uint16_t channels = 1;
+  if (read_stereo_) {
+    channels = 2;
+  }
+
+  int32_t payload_size = (int32_t) fread(audio_frame.data_, sizeof(uint16_t),
+                                         samples_10ms_ * channels, pcm_file_);
+  if (payload_size < samples_10ms_ * channels) {
+    for (int k = payload_size; k < samples_10ms_ * channels; k++) {
+      audio_frame.data_[k] = 0;
+    }
+    if (auto_rewind_) {
+      rewind(pcm_file_);
+      rewinded_ = true;
+    } else {
+      end_of_file_ = true;
+    }
+  }
+  audio_frame.samples_per_channel_ = samples_10ms_;
+  audio_frame.sample_rate_hz_ = frequency_;
+  audio_frame.num_channels_ = channels;
+  audio_frame.timestamp_ = timestamp_;
+  timestamp_ += samples_10ms_;
+  return samples_10ms_;
+}
+
+void PCMFile::Write10MsData(AudioFrame& audio_frame) {
+  if (audio_frame.num_channels_ == 1) {
+    if (!save_stereo_) {
+      if (fwrite(audio_frame.data_, sizeof(uint16_t),
+                 audio_frame.samples_per_channel_, pcm_file_) !=
+          static_cast<size_t>(audio_frame.samples_per_channel_)) {
+        return;
+      }
+    } else {
+      int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
+      for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
+        stereo_audio[k << 1] = audio_frame.data_[k];
+        stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
+      }
+      if (fwrite(stereo_audio, sizeof(int16_t),
+                 2 * audio_frame.samples_per_channel_, pcm_file_) !=
+          static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
+        return;
+      }
+      delete[] stereo_audio;
+    }
+  } else {
+    if (fwrite(audio_frame.data_, sizeof(int16_t),
+               audio_frame.num_channels_ * audio_frame.samples_per_channel_,
+               pcm_file_) !=
+        static_cast<size_t>(audio_frame.num_channels_ *
+                            audio_frame.samples_per_channel_)) {
+      return;
+    }
+  }
+}
+
+void PCMFile::Write10MsData(int16_t* playout_buffer, size_t length_smpls) {
+  if (fwrite(playout_buffer, sizeof(uint16_t), length_smpls, pcm_file_) !=
+      length_smpls) {
+    return;
+  }
+}
+
+void PCMFile::Close() {
+  fclose(pcm_file_);
+  pcm_file_ = NULL;
+}
+
+void PCMFile::Rewind() {
+  rewind(pcm_file_);
+  end_of_file_ = false;
+}
+
+bool PCMFile::Rewinded() {
+  return rewinded_;
+}
+
+void PCMFile::SaveStereo(bool is_stereo) {
+  save_stereo_ = is_stereo;
+}
+
+void PCMFile::ReadStereo(bool is_stereo) {
+  read_stereo_ = is_stereo;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/PCMFile.h

@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <string>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class PCMFile {
+ public:
+  PCMFile();
+  PCMFile(uint32_t timestamp);
+  ~PCMFile() {
+    if (pcm_file_ != NULL) {
+      fclose(pcm_file_);
+    }
+  }
+
+  void Open(const std::string& filename, uint16_t frequency, const char* mode,
+            bool auto_rewind = false);
+
+  int32_t Read10MsData(AudioFrame& audio_frame);
+
+  void Write10MsData(int16_t *playout_buffer, size_t length_smpls);
+  void Write10MsData(AudioFrame& audio_frame);
+
+  uint16_t PayloadLength10Ms() const;
+  int32_t SamplingFrequency() const;
+  void Close();
+  bool EndOfFile() const {
+    return end_of_file_;
+  }
+  void Rewind();
+  static int16_t ChooseFile(std::string* file_name, int16_t max_len,
+                            uint16_t* frequency_hz);
+  bool Rewinded();
+  void SaveStereo(bool is_stereo = true);
+  void ReadStereo(bool is_stereo = true);
+ private:
+  FILE* pcm_file_;
+  uint16_t samples_10ms_;
+  int32_t frequency_;
+  bool end_of_file_;
+  bool auto_rewind_;
+  bool rewinded_;
+  uint32_t timestamp_;
+  bool read_stereo_;
+  bool save_stereo_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_

webrtc/modules/audio_coding/test/PacketLossTest.cc

@@ -0,0 +1,167 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+ReceiverWithPacketLoss::ReceiverWithPacketLoss()
+    : loss_rate_(0),
+      burst_length_(1),
+      packet_counter_(0),
+      lost_packet_counter_(0),
+      burst_lost_counter_(burst_length_) {
+}
+
+void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
+                                   RTPStream *rtpStream,
+                                   std::string out_file_name,
+                                   int channels,
+                                   int loss_rate,
+                                   int burst_length) {
+  loss_rate_ = loss_rate;
+  burst_length_ = burst_length;
+  burst_lost_counter_ = burst_length_;  // To prevent first packet gets lost.
+  std::stringstream ss;
+  ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
+  Receiver::Setup(acm, rtpStream, ss.str(), channels);
+}
+
+bool ReceiverWithPacketLoss::IncomingPacket() {
+  if (!_rtpStream->EndOfFile()) {
+    if (packet_counter_ == 0) {
+      _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                               _payloadSizeBytes, &_nextTime);
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          packet_counter_ = 0;
+          return true;
+        } else {
+          return false;
+        }
+      }
+    }
+
+    if (!PacketLost()) {
+      _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
+    }
+    packet_counter_++;
+    _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
+                                             _payloadSizeBytes, &_nextTime);
+    if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
+      packet_counter_ = 0;
+      lost_packet_counter_ = 0;
+    }
+  }
+  return true;
+}
+
+bool ReceiverWithPacketLoss::PacketLost() {
+  if (burst_lost_counter_ < burst_length_) {
+    lost_packet_counter_++;
+    burst_lost_counter_++;
+    return true;
+  }
+
+  if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
+    lost_packet_counter_++;
+    burst_lost_counter_ = 1;
+    return true;
+  }
+  return false;
+}
+
+SenderWithFEC::SenderWithFEC()
+    : expected_loss_rate_(0) {
+}
+
+void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+                          std::string in_file_name, int sample_rate,
+                          int channels, int expected_loss_rate) {
+  Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
+  EXPECT_TRUE(SetFEC(true));
+  EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
+}
+
+bool SenderWithFEC::SetFEC(bool enable_fec) {
+  if (_acm->SetCodecFEC(enable_fec) == 0) {
+    return true;
+  }
+  return false;
+}
+
+bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
+  if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
+    expected_loss_rate_ = expected_loss_rate;
+    return true;
+  }
+  return false;
+}
+
+PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
+                               int actual_loss_rate, int burst_length)
+    : channels_(channels),
+      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
+                    "audio_coding/teststereo32kHz"),
+      sample_rate_hz_(32000),
+      sender_(new SenderWithFEC),
+      receiver_(new ReceiverWithPacketLoss),
+      expected_loss_rate_(expected_loss_rate),
+      actual_loss_rate_(actual_loss_rate),
+      burst_length_(burst_length) {
+}
+
+void PacketLossTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  return;
+#else
+  rtc::scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
+
+  int codec_id = acm->Codec("opus", 48000, channels_);
+
+  RTPFile rtpFile;
+  std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                    "packet_loss_test");
+
+  // Encode to file
+  rtpFile.Open(fileName.c_str(), "wb+");
+  rtpFile.WriteHeader();
+
+  sender_->testMode = 0;
+  sender_->codeId = codec_id;
+
+  sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
+                 expected_loss_rate_);
+  if (acm->SendCodec()) {
+    sender_->Run();
+  }
+  sender_->Teardown();
+  rtpFile.Close();
+
+  // Decode to file
+  rtpFile.Open(fileName.c_str(), "rb");
+  rtpFile.ReadHeader();
+
+  receiver_->testMode = 0;
+  receiver_->codeId = codec_id;
+
+  receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
+                   actual_loss_rate_, burst_length_);
+  receiver_->Run();
+  receiver_->Teardown();
+  rtpFile.Close();
+#endif
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/PacketLossTest.h

@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+
+#include <string>
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+
+namespace webrtc {
+
+class ReceiverWithPacketLoss : public Receiver {
+ public:
+  ReceiverWithPacketLoss();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string out_file_name, int channels, int loss_rate,
+             int burst_length);
+  bool IncomingPacket() override;
+
+ protected:
+  bool PacketLost();
+  int loss_rate_;
+  int burst_length_;
+  int packet_counter_;
+  int lost_packet_counter_;
+  int burst_lost_counter_;
+};
+
+class SenderWithFEC : public Sender {
+ public:
+  SenderWithFEC();
+  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
+             std::string in_file_name, int sample_rate, int channels,
+             int expected_loss_rate);
+  bool SetPacketLossRate(int expected_loss_rate);
+  bool SetFEC(bool enable_fec);
+ protected:
+  int expected_loss_rate_;
+};
+
+class PacketLossTest : public ACMTest {
+ public:
+  PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
+                 int burst_length);
+  void Perform();
+ protected:
+  int channels_;
+  std::string in_file_name_;
+  int sample_rate_hz_;
+  rtc::scoped_ptr<SenderWithFEC> sender_;
+  rtc::scoped_ptr<ReceiverWithPacketLoss> receiver_;
+  int expected_loss_rate_;
+  int actual_loss_rate_;
+  int burst_length_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_

webrtc/modules/audio_coding/test/RTPFile.cc

@@ -0,0 +1,227 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "RTPFile.h"
+
+#include <stdlib.h>
+#include <limits>
+
+#ifdef WIN32
+#   include <Winsock2.h>
+#else
+#   include <arpa/inet.h>
+#endif
+
+#include "audio_coding_module.h"
+#include "engine_configurations.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+// TODO(tlegrand): Consider removing usage of gtest.
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
+                               const uint8_t* rtpHeader) {
+  rtpInfo->header.payloadType = rtpHeader[1];
+  rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
+      rtpHeader[3];
+  rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
+      (static_cast<uint32_t>(rtpHeader[5]) << 16) |
+      (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
+  rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
+      (static_cast<uint32_t>(rtpHeader[9]) << 16) |
+      (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11];
+}
+
+void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
+                              int16_t seqNo, uint32_t timeStamp,
+                              uint32_t ssrc) {
+  rtpHeader[0] = 0x80;
+  rtpHeader[1] = payloadType;
+  rtpHeader[2] = (seqNo >> 8) & 0xFF;
+  rtpHeader[3] = seqNo & 0xFF;
+  rtpHeader[4] = timeStamp >> 24;
+  rtpHeader[5] = (timeStamp >> 16) & 0xFF;
+  rtpHeader[6] = (timeStamp >> 8) & 0xFF;
+  rtpHeader[7] = timeStamp & 0xFF;
+  rtpHeader[8] = ssrc >> 24;
+  rtpHeader[9] = (ssrc >> 16) & 0xFF;
+  rtpHeader[10] = (ssrc >> 8) & 0xFF;
+  rtpHeader[11] = ssrc & 0xFF;
+}
+
+RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
+                     const uint8_t* payloadData, size_t payloadSize,
+                     uint32_t frequency)
+    : payloadType(payloadType),
+      timeStamp(timeStamp),
+      seqNo(seqNo),
+      payloadSize(payloadSize),
+      frequency(frequency) {
+  if (payloadSize > 0) {
+    this->payloadData = new uint8_t[payloadSize];
+    memcpy(this->payloadData, payloadData, payloadSize);
+  }
+}
+
+RTPPacket::~RTPPacket() {
+  delete[] payloadData;
+}
+
+RTPBuffer::RTPBuffer() {
+  _queueRWLock = RWLockWrapper::CreateRWLock();
+}
+
+RTPBuffer::~RTPBuffer() {
+  delete _queueRWLock;
+}
+
+void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
+                      const int16_t seqNo, const uint8_t* payloadData,
+                      const size_t payloadSize, uint32_t frequency) {
+  RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
+                                    payloadSize, frequency);
+  _queueRWLock->AcquireLockExclusive();
+  _rtpQueue.push(packet);
+  _queueRWLock->ReleaseLockExclusive();
+}
+
+size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+                       size_t payloadSize, uint32_t* offset) {
+  _queueRWLock->AcquireLockShared();
+  RTPPacket *packet = _rtpQueue.front();
+  _rtpQueue.pop();
+  _queueRWLock->ReleaseLockShared();
+  rtpInfo->header.markerBit = 1;
+  rtpInfo->header.payloadType = packet->payloadType;
+  rtpInfo->header.sequenceNumber = packet->seqNo;
+  rtpInfo->header.ssrc = 0;
+  rtpInfo->header.timestamp = packet->timeStamp;
+  if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) {
+    memcpy(payloadData, packet->payloadData, packet->payloadSize);
+  } else {
+    return 0;
+  }
+  *offset = (packet->timeStamp / (packet->frequency / 1000));
+
+  return packet->payloadSize;
+}
+
+bool RTPBuffer::EndOfFile() const {
+  _queueRWLock->AcquireLockShared();
+  bool eof = _rtpQueue.empty();
+  _queueRWLock->ReleaseLockShared();
+  return eof;
+}
+
+void RTPFile::Open(const char *filename, const char *mode) {
+  if ((_rtpFile = fopen(filename, mode)) == NULL) {
+    printf("Cannot write file %s.\n", filename);
+    ADD_FAILURE() << "Unable to write file";
+    exit(1);
+  }
+}
+
+void RTPFile::Close() {
+  if (_rtpFile != NULL) {
+    fclose(_rtpFile);
+    _rtpFile = NULL;
+  }
+}
+
+void RTPFile::WriteHeader() {
+  // Write data in a format that NetEQ and RTP Play can parse
+  fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
+  uint32_t dummy_variable = 0;
+  // should be converted to network endian format, but does not matter when 0
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&dummy_variable, 2, 1, _rtpFile));
+  fflush(_rtpFile);
+}
+
+void RTPFile::ReadHeader() {
+  uint32_t start_sec, start_usec, source;
+  uint16_t port, padding;
+  char fileHeader[40];
+  EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
+  EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
+  start_sec = ntohl(start_sec);
+  EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
+  start_usec = ntohl(start_usec);
+  EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
+  source = ntohl(source);
+  EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
+  port = ntohs(port);
+  EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
+  padding = ntohs(padding);
+}
+
+void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
+                    const int16_t seqNo, const uint8_t* payloadData,
+                    const size_t payloadSize, uint32_t frequency) {
+  /* write RTP packet to file */
+  uint8_t rtpHeader[12];
+  MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
+  ASSERT_LE(12 + payloadSize + 8, std::numeric_limits<u_short>::max());
+  uint16_t lengthBytes = htons(static_cast<u_short>(12 + payloadSize + 8));
+  uint16_t plen = htons(static_cast<u_short>(12 + payloadSize));
+  uint32_t offsetMs;
+
+  offsetMs = (timeStamp / (frequency / 1000));
+  offsetMs = htonl(offsetMs);
+  EXPECT_EQ(1u, fwrite(&lengthBytes, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&plen, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&offsetMs, 4, 1, _rtpFile));
+  EXPECT_EQ(1u, fwrite(&rtpHeader, 12, 1, _rtpFile));
+  EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
+}
+
+size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+                     size_t payloadSize, uint32_t* offset) {
+  uint16_t lengthBytes;
+  uint16_t plen;
+  uint8_t rtpHeader[12];
+  size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
+  /* Check if we have reached end of file. */
+  if ((read_len == 0) && feof(_rtpFile)) {
+    _rtpEOF = true;
+    return 0;
+  }
+  EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
+  EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
+  lengthBytes = ntohs(lengthBytes);
+  plen = ntohs(plen);
+  *offset = ntohl(*offset);
+  EXPECT_GT(plen, 11);
+
+  EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
+  ParseRTPHeader(rtpInfo, rtpHeader);
+  rtpInfo->type.Audio.isCNG = false;
+  rtpInfo->type.Audio.channel = 1;
+  EXPECT_EQ(lengthBytes, plen + 8);
+
+  if (plen == 0) {
+    return 0;
+  }
+  if (lengthBytes < 20) {
+    return 0;
+  }
+  if (payloadSize < static_cast<size_t>((lengthBytes - 20))) {
+    return 0;
+  }
+  lengthBytes -= 20;
+  EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
+  return lengthBytes;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/RTPFile.h

@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+
+#include <stdio.h>
+#include <queue>
+
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class RTPStream {
+ public:
+  virtual ~RTPStream() {
+  }
+
+  virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
+                     const int16_t seqNo, const uint8_t* payloadData,
+                     const size_t payloadSize, uint32_t frequency) = 0;
+
+  // Returns the packet's payload size. Zero should be treated as an
+  // end-of-stream (in the case that EndOfFile() is true) or an error.
+  virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
+                      size_t payloadSize, uint32_t* offset) = 0;
+  virtual bool EndOfFile() const = 0;
+
+ protected:
+  void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
+                     uint32_t timeStamp, uint32_t ssrc);
+
+  void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
+};
+
+class RTPPacket {
+ public:
+  RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
+            const uint8_t* payloadData, size_t payloadSize,
+            uint32_t frequency);
+
+  ~RTPPacket();
+
+  uint8_t payloadType;
+  uint32_t timeStamp;
+  int16_t seqNo;
+  uint8_t* payloadData;
+  size_t payloadSize;
+  uint32_t frequency;
+};
+
+class RTPBuffer : public RTPStream {
+ public:
+  RTPBuffer();
+
+  ~RTPBuffer();
+
+  void Write(const uint8_t payloadType,
+             const uint32_t timeStamp,
+             const int16_t seqNo,
+             const uint8_t* payloadData,
+             const size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(WebRtcRTPHeader* rtpInfo,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override;
+
+ private:
+  RWLockWrapper* _queueRWLock;
+  std::queue<RTPPacket *> _rtpQueue;
+};
+
+class RTPFile : public RTPStream {
+ public:
+  ~RTPFile() {
+  }
+
+  RTPFile()
+      : _rtpFile(NULL),
+        _rtpEOF(false) {
+  }
+
+  void Open(const char *outFilename, const char *mode);
+
+  void Close();
+
+  void WriteHeader();
+
+  void ReadHeader();
+
+  void Write(const uint8_t payloadType,
+             const uint32_t timeStamp,
+             const int16_t seqNo,
+             const uint8_t* payloadData,
+             const size_t payloadSize,
+             uint32_t frequency) override;
+
+  size_t Read(WebRtcRTPHeader* rtpInfo,
+              uint8_t* payloadData,
+              size_t payloadSize,
+              uint32_t* offset) override;
+
+  bool EndOfFile() const override { return _rtpEOF; }
+
+ private:
+  FILE* _rtpFile;
+  bool _rtpEOF;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_

webrtc/modules/audio_coding/test/SpatialAudio.cc

@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string.h>
+
+#include <math.h>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/test/SpatialAudio.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#define NUM_PANN_COEFFS 10
+
+SpatialAudio::SpatialAudio(int testMode)
+    : _acmLeft(AudioCodingModule::Create(1)),
+      _acmRight(AudioCodingModule::Create(2)),
+      _acmReceiver(AudioCodingModule::Create(3)),
+      _testMode(testMode) {
+}
+
+SpatialAudio::~SpatialAudio() {
+  delete _channel;
+  _inFile.Close();
+  _outFile.Close();
+}
+
+int16_t SpatialAudio::Setup() {
+  _channel = new Channel;
+
+  // Register callback for the sender side.
+  CHECK_ERROR(_acmLeft->RegisterTransportCallback(_channel));
+  CHECK_ERROR(_acmRight->RegisterTransportCallback(_channel));
+  // Register the receiver ACM in channel
+  _channel->RegisterReceiverACM(_acmReceiver.get());
+
+  uint16_t sampFreqHz = 32000;
+
+  const std::string file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  _inFile.Open(file_name, sampFreqHz, "rb", false);
+
+  std::string output_file = webrtc::test::OutputPath()
+      + "out_spatial_autotest.pcm";
+  if (_testMode == 1) {
+    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
+    printf("\n");
+    printf("Enter the output file [%s]: ", output_file.c_str());
+    PCMFile::ChooseFile(&output_file, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
+  } else {
+    output_file = webrtc::test::OutputPath() + "testspatial_out.pcm";
+  }
+  _outFile.Open(output_file, sampFreqHz, "wb", false);
+  _outFile.SaveStereo(true);
+
+  // Register all available codes as receiving codecs.
+  CodecInst codecInst;
+  int status;
+  uint8_t num_encoders = _acmReceiver->NumberOfCodecs();
+  // Register all available codes as receiving codecs once more.
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    status = _acmReceiver->Codec(n, &codecInst);
+    if (status < 0) {
+      printf("Error in Codec(), no matching codec found");
+    }
+    status = _acmReceiver->RegisterReceiveCodec(codecInst);
+    if (status < 0) {
+      printf("Error in RegisterReceiveCodec() for payload type %d",
+             codecInst.pltype);
+    }
+  }
+
+  return 0;
+}
+
+void SpatialAudio::Perform() {
+  if (_testMode == 0) {
+    printf("Running SpatialAudio Test");
+    WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
+                 "---------- SpatialAudio ----------");
+  }
+
+  Setup();
+
+  CodecInst codecInst;
+  _acmLeft->Codec((uint8_t) 1, &codecInst);
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  EncodeDecode();
+
+  int16_t pannCntr = 0;
+
+  double leftPanning[NUM_PANN_COEFFS] = { 1.00, 0.95, 0.90, 0.85, 0.80, 0.75,
+      0.70, 0.60, 0.55, 0.50 };
+  double rightPanning[NUM_PANN_COEFFS] = { 0.50, 0.55, 0.60, 0.70, 0.75, 0.80,
+      0.85, 0.90, 0.95, 1.00 };
+
+  while ((pannCntr + 1) < NUM_PANN_COEFFS) {
+    _acmLeft->Codec((uint8_t) 0, &codecInst);
+    codecInst.pacsize = 480;
+    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr++;
+
+    // Change codec
+    _acmLeft->Codec((uint8_t) 3, &codecInst);
+    codecInst.pacsize = 320;
+    CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+    CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr++;
+    if (_testMode == 0) {
+      printf(".");
+    }
+  }
+
+  _acmLeft->Codec((uint8_t) 4, &codecInst);
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  EncodeDecode();
+
+  _acmLeft->Codec((uint8_t) 0, &codecInst);
+  codecInst.pacsize = 480;
+  CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
+  CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
+  pannCntr = NUM_PANN_COEFFS - 1;
+  while (pannCntr >= 0) {
+    EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
+    pannCntr--;
+    if (_testMode == 0) {
+      printf(".");
+    }
+  }
+  if (_testMode == 0) {
+    printf("Done!\n");
+  }
+}
+
+void SpatialAudio::EncodeDecode(const double leftPanning,
+                                const double rightPanning) {
+  AudioFrame audioFrame;
+  int32_t outFileSampFreq = _outFile.SamplingFrequency();
+
+  const double rightToLeftRatio = rightPanning / leftPanning;
+
+  _channel->SetIsStereo(true);
+
+  while (!_inFile.EndOfFile()) {
+    _inFile.Read10MsData(audioFrame);
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
+      audioFrame.data_[n] = (int16_t) floor(
+          audioFrame.data_[n] * leftPanning + 0.5);
+    }
+    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+    for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
+      audioFrame.data_[n] = (int16_t) floor(
+          audioFrame.data_[n] * rightToLeftRatio + 0.5);
+    }
+    CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
+
+    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
+    _outFile.Write10MsData(audioFrame);
+  }
+  _inFile.Rewind();
+}
+
+void SpatialAudio::EncodeDecode() {
+  AudioFrame audioFrame;
+  int32_t outFileSampFreq = _outFile.SamplingFrequency();
+
+  _channel->SetIsStereo(false);
+
+  while (!_inFile.EndOfFile()) {
+    _inFile.Read10MsData(audioFrame);
+    CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
+
+    CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, &audioFrame));
+    _outFile.Write10MsData(audioFrame);
+  }
+  _inFile.Rewind();
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/SpatialAudio.h

@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+namespace webrtc {
+
+class SpatialAudio : public ACMTest {
+ public:
+  SpatialAudio(int testMode);
+  ~SpatialAudio();
+
+  void Perform();
+ private:
+  int16_t Setup();
+  void EncodeDecode(double leftPanning, double rightPanning);
+  void EncodeDecode();
+
+  rtc::scoped_ptr<AudioCodingModule> _acmLeft;
+  rtc::scoped_ptr<AudioCodingModule> _acmRight;
+  rtc::scoped_ptr<AudioCodingModule> _acmReceiver;
+  Channel* _channel;
+  PCMFile _inFile;
+  PCMFile _outFile;
+  int _testMode;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_

webrtc/modules/audio_coding/test/TestAllCodecs.cc

@@ -0,0 +1,485 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+
+#include <cstdio>
+#include <limits>
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/typedefs.h"
+
+// Description of the test:
+// In this test we set up a one-way communication channel from a participant
+// called "a" to a participant called "b".
+// a -> channel_a_to_b -> b
+//
+// The test loops through all available mono codecs, encode at "a" sends over
+// the channel, and decodes at "b".
+
+namespace {
+const size_t kVariableSize = std::numeric_limits<size_t>::max();
+}
+
+namespace webrtc {
+
+// Class for simulating packet handling.
+TestPack::TestPack()
+    : receiver_acm_(NULL),
+      sequence_number_(0),
+      timestamp_diff_(0),
+      last_in_timestamp_(0),
+      total_bytes_(0),
+      payload_size_(0) {
+}
+
+TestPack::~TestPack() {
+}
+
+void TestPack::RegisterReceiverACM(AudioCodingModule* acm) {
+  receiver_acm_ = acm;
+  return;
+}
+
+int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
+                           uint32_t timestamp, const uint8_t* payload_data,
+                           size_t payload_size,
+                           const RTPFragmentationHeader* fragmentation) {
+  WebRtcRTPHeader rtp_info;
+  int32_t status;
+
+  rtp_info.header.markerBit = false;
+  rtp_info.header.ssrc = 0;
+  rtp_info.header.sequenceNumber = sequence_number_++;
+  rtp_info.header.payloadType = payload_type;
+  rtp_info.header.timestamp = timestamp;
+  if (frame_type == kAudioFrameCN) {
+    rtp_info.type.Audio.isCNG = true;
+  } else {
+    rtp_info.type.Audio.isCNG = false;
+  }
+  if (frame_type == kEmptyFrame) {
+    // Skip this frame.
+    return 0;
+  }
+
+  // Only run mono for all test cases.
+  rtp_info.type.Audio.channel = 1;
+  memcpy(payload_data_, payload_data, payload_size);
+
+  status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
+
+  payload_size_ = payload_size;
+  timestamp_diff_ = timestamp - last_in_timestamp_;
+  last_in_timestamp_ = timestamp;
+  total_bytes_ += payload_size;
+  return status;
+}
+
+size_t TestPack::payload_size() {
+  return payload_size_;
+}
+
+uint32_t TestPack::timestamp_diff() {
+  return timestamp_diff_;
+}
+
+void TestPack::reset_payload_size() {
+  payload_size_ = 0;
+}
+
+TestAllCodecs::TestAllCodecs(int test_mode)
+    : acm_a_(AudioCodingModule::Create(0)),
+      acm_b_(AudioCodingModule::Create(1)),
+      channel_a_to_b_(NULL),
+      test_count_(0),
+      packet_size_samples_(0),
+      packet_size_bytes_(0) {
+  // test_mode = 0 for silent test (auto test)
+  test_mode_ = test_mode;
+}
+
+TestAllCodecs::~TestAllCodecs() {
+  if (channel_a_to_b_ != NULL) {
+    delete channel_a_to_b_;
+    channel_a_to_b_ = NULL;
+  }
+}
+
+void TestAllCodecs::Perform() {
+  const std::string file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  infile_a_.Open(file_name, 32000, "rb");
+
+  if (test_mode_ == 0) {
+    WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
+                 "---------- TestAllCodecs ----------");
+  }
+
+  acm_a_->InitializeReceiver();
+  acm_b_->InitializeReceiver();
+
+  uint8_t num_encoders = acm_a_->NumberOfCodecs();
+  CodecInst my_codec_param;
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    acm_b_->Codec(n, &my_codec_param);
+    if (!strcmp(my_codec_param.plname, "opus")) {
+      my_codec_param.channels = 1;
+    }
+    acm_b_->RegisterReceiveCodec(my_codec_param);
+  }
+
+  // Create and connect the channel
+  channel_a_to_b_ = new TestPack;
+  acm_a_->RegisterTransportCallback(channel_a_to_b_);
+  channel_a_to_b_->RegisterReceiverACM(acm_b_.get());
+
+  // All codecs are tested for all allowed sampling frequencies, rates and
+  // packet sizes.
+#ifdef WEBRTC_CODEC_G722
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_g722[] = "G722";
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 640, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 800, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_ilbc[] = "ILBC";
+  RegisterSendCodec('A', codec_ilbc, 8000, 13300, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 13300, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 15200, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_isac[] = "ISAC";
+  RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_l16[] = "L16";
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 320, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 480, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 640, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 640, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_pcma[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  char codec_pcmu[] = "PCMU";
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, 0);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, 0);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#ifdef WEBRTC_CODEC_OPUS
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+  }
+  test_count_++;
+  OpenOutFile(test_count_);
+  char codec_opus[] = "OPUS";
+  RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 20000, 480*2, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 480*4, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 96000, 480*6, kVariableSize);
+  Run(channel_a_to_b_);
+  RegisterSendCodec('A', codec_opus, 48000, 500000, 480*2, kVariableSize);
+  Run(channel_a_to_b_);
+  outfile_b_.Close();
+#endif
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+
+    /* Print out all codecs that were not tested in the run */
+    printf("The following codecs was not included in the test:\n");
+#ifndef WEBRTC_CODEC_G722
+    printf("   G.722\n");
+#endif
+#ifndef WEBRTC_CODEC_ILBC
+    printf("   iLBC\n");
+#endif
+#ifndef WEBRTC_CODEC_ISAC
+    printf("   ISAC float\n");
+#endif
+#ifndef WEBRTC_CODEC_ISACFX
+    printf("   ISAC fix\n");
+#endif
+
+    printf("\nTo complete the test, listen to the %d number of output files.\n",
+           test_count_);
+  }
+}
+
+// Register Codec to use in the test
+//
+// Input:  side             - which ACM to use, 'A' or 'B'
+//         codec_name       - name to use when register the codec
+//         sampling_freq_hz - sampling frequency in Herz
+//         rate             - bitrate in bytes
+//         packet_size      - packet size in samples
+//         extra_byte       - if extra bytes needed compared to the bitrate
+//                            used when registering, can be an internal header
+//                            set to kVariableSize if the codec is a variable
+//                            rate codec
+void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
+                                      int32_t sampling_freq_hz, int rate,
+                                      int packet_size, size_t extra_byte) {
+  if (test_mode_ != 0) {
+    // Print out codec and settings.
+    printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
+           sampling_freq_hz, rate, packet_size);
+  }
+
+  // Store packet-size in samples, used to validate the received packet.
+  // If G.722, store half the size to compensate for the timestamp bug in the
+  // RFC for G.722.
+  // If iSAC runs in adaptive mode, packet size in samples can change on the
+  // fly, so we exclude this test by setting |packet_size_samples_| to -1.
+  if (!strcmp(codec_name, "G722")) {
+    packet_size_samples_ = packet_size / 2;
+  } else if (!strcmp(codec_name, "ISAC") && (rate == -1)) {
+    packet_size_samples_ = -1;
+  } else {
+    packet_size_samples_ = packet_size;
+  }
+
+  // Store the expected packet size in bytes, used to validate the received
+  // packet. If variable rate codec (extra_byte == -1), set to -1.
+  if (extra_byte != kVariableSize) {
+    // Add 0.875 to always round up to a whole byte
+    packet_size_bytes_ = static_cast<size_t>(
+        static_cast<float>(packet_size * rate) /
+        static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
+  } else {
+    // Packets will have a variable size.
+    packet_size_bytes_ = kVariableSize;
+  }
+
+  // Set pointer to the ACM where to register the codec.
+  AudioCodingModule* my_acm = NULL;
+  switch (side) {
+    case 'A': {
+      my_acm = acm_a_.get();
+      break;
+    }
+    case 'B': {
+      my_acm = acm_b_.get();
+      break;
+    }
+    default: {
+      break;
+    }
+  }
+  ASSERT_TRUE(my_acm != NULL);
+
+  // Get all codec parameters before registering
+  CodecInst my_codec_param;
+  CHECK_ERROR(AudioCodingModule::Codec(codec_name, &my_codec_param,
+                                       sampling_freq_hz, 1));
+  my_codec_param.rate = rate;
+  my_codec_param.pacsize = packet_size;
+  CHECK_ERROR(my_acm->RegisterSendCodec(my_codec_param));
+}
+
+void TestAllCodecs::Run(TestPack* channel) {
+  AudioFrame audio_frame;
+
+  int32_t out_freq_hz = outfile_b_.SamplingFrequency();
+  size_t receive_size;
+  uint32_t timestamp_diff;
+  channel->reset_payload_size();
+  int error_count = 0;
+
+  int counter = 0;
+  while (!infile_a_.EndOfFile()) {
+    // Add 10 msec to ACM.
+    infile_a_.Read10MsData(audio_frame);
+    CHECK_ERROR(acm_a_->Add10MsData(audio_frame));
+
+    // Verify that the received packet size matches the settings.
+    receive_size = channel->payload_size();
+    if (receive_size) {
+      if ((receive_size != packet_size_bytes_) &&
+          (packet_size_bytes_ != kVariableSize)) {
+        error_count++;
+      }
+
+      // Verify that the timestamp is updated with expected length. The counter
+      // is used to avoid problems when switching codec or frame size in the
+      // test.
+      timestamp_diff = channel->timestamp_diff();
+      if ((counter > 10) &&
+          (static_cast<int>(timestamp_diff) != packet_size_samples_) &&
+          (packet_size_samples_ > -1))
+        error_count++;
+    }
+
+    // Run received side of ACM.
+    CHECK_ERROR(acm_b_->PlayoutData10Ms(out_freq_hz, &audio_frame));
+
+    // Write output speech to file.
+    outfile_b_.Write10MsData(audio_frame.data_,
+                             audio_frame.samples_per_channel_);
+
+    // Update loop counter
+    counter++;
+  }
+
+  EXPECT_EQ(0, error_count);
+
+  if (infile_a_.EndOfFile()) {
+    infile_a_.Rewind();
+  }
+}
+
+void TestAllCodecs::OpenOutFile(int test_number) {
+  std::string filename = webrtc::test::OutputPath();
+  std::ostringstream test_number_str;
+  test_number_str << test_number;
+  filename += "testallcodecs_out_";
+  filename += test_number_str.str();
+  filename += ".pcm";
+  outfile_b_.Open(filename, 32000, "wb");
+}
+
+void TestAllCodecs::DisplaySendReceiveCodec() {
+  CodecInst my_codec_param;
+  printf("%s -> ", acm_a_->SendCodec()->plname);
+  acm_b_->ReceiveCodec(&my_codec_param);
+  printf("%s\n", my_codec_param.plname);
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/TestAllCodecs.h

@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Config;
+
+class TestPack : public AudioPacketizationCallback {
+ public:
+  TestPack();
+  ~TestPack();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(FrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  size_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  uint16_t sequence_number_;
+  uint8_t payload_data_[60 * 32 * 2 * 2];
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  size_t payload_size_;
+};
+
+class TestAllCodecs : public ACMTest {
+ public:
+  explicit TestAllCodecs(int test_mode);
+  ~TestAllCodecs();
+
+  void Perform() override;
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name, and a sampling frequency matching is not required.
+  // This is useful for codecs which support several sampling frequency.
+  // Note! Only mono mode is tested in this test.
+  void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
+                         int rate, int packet_size, size_t extra_byte);
+
+  void Run(TestPack* channel);
+  void OpenOutFile(int test_number);
+  void DisplaySendReceiveCodec();
+
+  int test_mode_;
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  TestPack* channel_a_to_b_;
+  PCMFile infile_a_;
+  PCMFile outfile_b_;
+  int test_count_;
+  int packet_size_samples_;
+  size_t packet_size_bytes_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_

webrtc/modules/audio_coding/test/TestRedFec.cc

@@ -0,0 +1,476 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+
+#include <assert.h>
+
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+#ifdef SUPPORT_RED_WB
+#undef SUPPORT_RED_WB
+#endif
+
+#ifdef SUPPORT_RED_SWB
+#undef SUPPORT_RED_SWB
+#endif
+
+#ifdef SUPPORT_RED_FB
+#undef SUPPORT_RED_FB
+#endif
+
+namespace webrtc {
+
+namespace {
+  const char kNameL16[] = "L16";
+  const char kNamePCMU[] = "PCMU";
+  const char kNameCN[] = "CN";
+  const char kNameRED[] = "RED";
+
+  // These three are only used by code #ifdeffed on WEBRTC_CODEC_G722.
+#ifdef WEBRTC_CODEC_G722
+  const char kNameISAC[] = "ISAC";
+  const char kNameG722[] = "G722";
+  const char kNameOPUS[] = "opus";
+#endif
+}
+
+TestRedFec::TestRedFec()
+    : _acmA(AudioCodingModule::Create(0)),
+      _acmB(AudioCodingModule::Create(1)),
+      _channelA2B(NULL),
+      _testCntr(0) {
+}
+
+TestRedFec::~TestRedFec() {
+  if (_channelA2B != NULL) {
+    delete _channelA2B;
+    _channelA2B = NULL;
+  }
+}
+
+void TestRedFec::Perform() {
+  const std::string file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  _inFileA.Open(file_name, 32000, "rb");
+
+  ASSERT_EQ(0, _acmA->InitializeReceiver());
+  ASSERT_EQ(0, _acmB->InitializeReceiver());
+
+  uint8_t numEncoders = _acmA->NumberOfCodecs();
+  CodecInst myCodecParam;
+  for (uint8_t n = 0; n < numEncoders; n++) {
+    EXPECT_EQ(0, _acmB->Codec(n, &myCodecParam));
+    // Default number of channels is 2 for opus, so we change to 1 in this test.
+    if (!strcmp(myCodecParam.plname, "opus")) {
+      myCodecParam.channels = 1;
+    }
+    EXPECT_EQ(0, _acmB->RegisterReceiveCodec(myCodecParam));
+  }
+
+  // Create and connect the channel
+  _channelA2B = new Channel;
+  _acmA->RegisterTransportCallback(_channelA2B);
+  _channelA2B->RegisterReceiverACM(_acmB.get());
+
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameL16, 8000));
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 8000));
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameRED));
+  EXPECT_EQ(0, SetVAD(true, true, VADAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNamePCMU, 8000);
+  // Switch to another 8 kHz codec, RED should remain switched on.
+  EXPECT_TRUE(_acmA->REDStatus());
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+#ifndef WEBRTC_CODEC_G722
+  EXPECT_TRUE(false);
+  printf("G722 needs to be activated to run this test\n");
+  return;
+#else
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
+
+#ifdef SUPPORT_RED_WB
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  // Switch to a 16 kHz codec, RED should have been switched off.
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+  Run();
+#ifdef SUPPORT_RED_WB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+
+#ifdef SUPPORT_RED_WB
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+
+#ifdef SUPPORT_RED_WB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+
+#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  // Switch to a 32 kHz codec, RED should have been switched off.
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+
+#ifdef SUPPORT_RED_SWB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+  EXPECT_EQ(0, SetVAD(false, false, VADNormal));
+
+#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  _channelA2B->SetFECTestWithPacketLoss(true);
+  // Following tests are under packet losses.
+
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameG722));
+  EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
+
+#if defined(SUPPORT_RED_WB) && defined(SUPPORT_RED_SWB)
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  // Switch to a 16 kHz codec, RED should have been switched off.
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+
+#ifdef SUPPORT_RED_WB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+
+#ifdef SUPPORT_RED_WB
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  // Switch to a 16 kHz codec, RED should have been switched off.
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+#ifdef SUPPORT_RED_WB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+
+#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
+  // Switch codec, RED should remain.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  // Switch to a 32 kHz codec, RED should have been switched off.
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_FALSE(_acmA->REDStatus());
+#ifdef SUPPORT_RED_SWB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  OpenOutFile(_testCntr);
+  Run();
+  _outFileB.Close();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+  EXPECT_EQ(0, SetVAD(false, false, VADNormal));
+#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
+  OpenOutFile(_testCntr);
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 32000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+  _outFileB.Close();
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+#ifndef WEBRTC_CODEC_OPUS
+  EXPECT_TRUE(false);
+  printf("Opus needs to be activated to run this test\n");
+  return;
+#endif
+
+  RegisterSendCodec('A', kNameOPUS, 48000);
+
+#if defined(SUPPORT_RED_FB) && defined(SUPPORT_RED_SWB) &&\
+  defined(SUPPORT_RED_WB)
+  // Switch to codec, RED should remain switched on.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+
+  // _channelA2B imposes 25% packet loss rate.
+  EXPECT_EQ(0, _acmA->SetPacketLossRate(25));
+
+#ifdef SUPPORT_RED_FB
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+  // Codec FEC and RED are mutually exclusive.
+  EXPECT_EQ(-1, _acmA->SetCodecFEC(true));
+
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_EQ(0, _acmA->SetCodecFEC(true));
+
+  // Codec FEC and RED are mutually exclusive.
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+#else
+  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
+  EXPECT_FALSE(_acmA->REDStatus());
+  EXPECT_EQ(0, _acmA->SetCodecFEC(true));
+#endif
+
+  EXPECT_TRUE(_acmA->CodecFEC());
+  OpenOutFile(_testCntr);
+  Run();
+
+  // Switch to L16 with RED.
+  RegisterSendCodec('A', kNameL16, 8000);
+  EXPECT_EQ(0, SetVAD(false, false, VADNormal));
+
+  // L16 does not support FEC, so FEC should be turned off automatically.
+  EXPECT_FALSE(_acmA->CodecFEC());
+
+  EXPECT_EQ(0, _acmA->SetREDStatus(true));
+  EXPECT_TRUE(_acmA->REDStatus());
+  Run();
+
+  // Switch to Opus again.
+  RegisterSendCodec('A', kNameOPUS, 48000);
+#ifdef SUPPORT_RED_FB
+  // Switch to codec, RED should remain switched on.
+  EXPECT_TRUE(_acmA->REDStatus());
+#else
+  EXPECT_FALSE(_acmA->REDStatus());
+#endif
+  EXPECT_EQ(0, _acmA->SetREDStatus(false));
+  EXPECT_EQ(0, _acmA->SetCodecFEC(false));
+  Run();
+
+  EXPECT_EQ(0, _acmA->SetCodecFEC(true));
+  _outFileB.Close();
+
+  // Codecs does not support internal FEC, cannot enable FEC.
+  RegisterSendCodec('A', kNameG722, 16000);
+  EXPECT_FALSE(_acmA->REDStatus());
+  EXPECT_EQ(-1, _acmA->SetCodecFEC(true));
+  EXPECT_FALSE(_acmA->CodecFEC());
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_FALSE(_acmA->REDStatus());
+  EXPECT_EQ(-1, _acmA->SetCodecFEC(true));
+  EXPECT_FALSE(_acmA->CodecFEC());
+
+  // Codecs does not support internal FEC, disable FEC does not trigger failure.
+  RegisterSendCodec('A', kNameG722, 16000);
+  EXPECT_FALSE(_acmA->REDStatus());
+  EXPECT_EQ(0, _acmA->SetCodecFEC(false));
+  EXPECT_FALSE(_acmA->CodecFEC());
+
+  RegisterSendCodec('A', kNameISAC, 16000);
+  EXPECT_FALSE(_acmA->REDStatus());
+  EXPECT_EQ(0, _acmA->SetCodecFEC(false));
+  EXPECT_FALSE(_acmA->CodecFEC());
+
+#endif  // defined(WEBRTC_CODEC_G722)
+}
+
+int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
+  return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
+}
+
+int16_t TestRedFec::RegisterSendCodec(char side, const char* codecName,
+                                      int32_t samplingFreqHz) {
+  std::cout << std::flush;
+  AudioCodingModule* myACM;
+  switch (side) {
+    case 'A': {
+      myACM = _acmA.get();
+      break;
+    }
+    case 'B': {
+      myACM = _acmB.get();
+      break;
+    }
+    default:
+      return -1;
+  }
+
+  if (myACM == NULL) {
+    assert(false);
+    return -1;
+  }
+  CodecInst myCodecParam;
+  EXPECT_GT(AudioCodingModule::Codec(codecName, &myCodecParam,
+                                     samplingFreqHz, 1), -1);
+  EXPECT_GT(myACM->RegisterSendCodec(myCodecParam), -1);
+
+  // Initialization was successful.
+  return 0;
+}
+
+void TestRedFec::Run() {
+  AudioFrame audioFrame;
+  int32_t outFreqHzB = _outFileB.SamplingFrequency();
+
+  while (!_inFileA.EndOfFile()) {
+    EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+    EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    _outFileB.Write10MsData(audioFrame.data_, audioFrame.samples_per_channel_);
+  }
+  _inFileA.Rewind();
+}
+
+void TestRedFec::OpenOutFile(int16_t test_number) {
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath();
+  file_stream << "TestRedFec_outFile_";
+  file_stream << test_number << ".pcm";
+  file_name = file_stream.str();
+  _outFileB.Open(file_name, 16000, "wb");
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/TestRedFec.h

@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+
+#include <string>
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+
+namespace webrtc {
+
+class Config;
+
+class TestRedFec : public ACMTest {
+ public:
+  explicit TestRedFec();
+  ~TestRedFec();
+
+  void Perform();
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name and a sampling frequency matching is not required. This is
+  // useful for codecs which support several sampling frequency.
+  int16_t RegisterSendCodec(char side, const char* codecName,
+                            int32_t sampFreqHz = -1);
+  void Run();
+  void OpenOutFile(int16_t testNumber);
+  int32_t SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
+  rtc::scoped_ptr<AudioCodingModule> _acmA;
+  rtc::scoped_ptr<AudioCodingModule> _acmB;
+
+  Channel* _channelA2B;
+
+  PCMFile _inFileA;
+  PCMFile _outFileB;
+  int16_t _testCntr;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_

webrtc/modules/audio_coding/test/TestStereo.cc

@@ -0,0 +1,838 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+
+#include <assert.h>
+
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+// Class for simulating packet handling
+TestPackStereo::TestPackStereo()
+    : receiver_acm_(NULL),
+      seq_no_(0),
+      timestamp_diff_(0),
+      last_in_timestamp_(0),
+      total_bytes_(0),
+      payload_size_(0),
+      codec_mode_(kNotSet),
+      lost_packet_(false) {
+}
+
+TestPackStereo::~TestPackStereo() {
+}
+
+void TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm) {
+  receiver_acm_ = acm;
+  return;
+}
+
+int32_t TestPackStereo::SendData(const FrameType frame_type,
+                                 const uint8_t payload_type,
+                                 const uint32_t timestamp,
+                                 const uint8_t* payload_data,
+                                 const size_t payload_size,
+                                 const RTPFragmentationHeader* fragmentation) {
+  WebRtcRTPHeader rtp_info;
+  int32_t status = 0;
+
+  rtp_info.header.markerBit = false;
+  rtp_info.header.ssrc = 0;
+  rtp_info.header.sequenceNumber = seq_no_++;
+  rtp_info.header.payloadType = payload_type;
+  rtp_info.header.timestamp = timestamp;
+  if (frame_type == kEmptyFrame) {
+    // Skip this frame
+    return 0;
+  }
+
+  if (lost_packet_ == false) {
+    if (frame_type != kAudioFrameCN) {
+      rtp_info.type.Audio.isCNG = false;
+      rtp_info.type.Audio.channel = static_cast<int>(codec_mode_);
+    } else {
+      rtp_info.type.Audio.isCNG = true;
+      rtp_info.type.Audio.channel = static_cast<int>(kMono);
+    }
+    status = receiver_acm_->IncomingPacket(payload_data, payload_size,
+                                           rtp_info);
+
+    if (frame_type != kAudioFrameCN) {
+      payload_size_ = static_cast<int>(payload_size);
+    } else {
+      payload_size_ = -1;
+    }
+
+    timestamp_diff_ = timestamp - last_in_timestamp_;
+    last_in_timestamp_ = timestamp;
+    total_bytes_ += payload_size;
+  }
+  return status;
+}
+
+uint16_t TestPackStereo::payload_size() {
+  return static_cast<uint16_t>(payload_size_);
+}
+
+uint32_t TestPackStereo::timestamp_diff() {
+  return timestamp_diff_;
+}
+
+void TestPackStereo::reset_payload_size() {
+  payload_size_ = 0;
+}
+
+void TestPackStereo::set_codec_mode(enum StereoMonoMode mode) {
+  codec_mode_ = mode;
+}
+
+void TestPackStereo::set_lost_packet(bool lost) {
+  lost_packet_ = lost;
+}
+
+TestStereo::TestStereo(int test_mode)
+    : acm_a_(AudioCodingModule::Create(0)),
+      acm_b_(AudioCodingModule::Create(1)),
+      channel_a2b_(NULL),
+      test_cntr_(0),
+      pack_size_samp_(0),
+      pack_size_bytes_(0),
+      counter_(0)
+#ifdef WEBRTC_CODEC_G722
+      , g722_pltype_(0)
+#endif
+      , l16_8khz_pltype_(-1)
+      , l16_16khz_pltype_(-1)
+      , l16_32khz_pltype_(-1)
+#ifdef PCMA_AND_PCMU
+      , pcma_pltype_(-1)
+      , pcmu_pltype_(-1)
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+      , opus_pltype_(-1)
+#endif
+      {
+  // test_mode = 0 for silent test (auto test)
+  test_mode_ = test_mode;
+}
+
+TestStereo::~TestStereo() {
+  if (channel_a2b_ != NULL) {
+    delete channel_a2b_;
+    channel_a2b_ = NULL;
+  }
+}
+
+void TestStereo::Perform() {
+  uint16_t frequency_hz;
+  int audio_channels;
+  int codec_channels;
+  bool dtx;
+  bool vad;
+  ACMVADMode vad_mode;
+
+  // Open both mono and stereo test files in 32 kHz.
+  const std::string file_name_stereo = webrtc::test::ResourcePath(
+      "audio_coding/teststereo32kHz", "pcm");
+  const std::string file_name_mono = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  frequency_hz = 32000;
+  in_file_stereo_ = new PCMFile();
+  in_file_mono_ = new PCMFile();
+  in_file_stereo_->Open(file_name_stereo, frequency_hz, "rb");
+  in_file_stereo_->ReadStereo(true);
+  in_file_mono_->Open(file_name_mono, frequency_hz, "rb");
+  in_file_mono_->ReadStereo(false);
+
+  // Create and initialize two ACMs, one for each side of a one-to-one call.
+  ASSERT_TRUE((acm_a_.get() != NULL) && (acm_b_.get() != NULL));
+  EXPECT_EQ(0, acm_a_->InitializeReceiver());
+  EXPECT_EQ(0, acm_b_->InitializeReceiver());
+
+  // Register all available codes as receiving codecs.
+  uint8_t num_encoders = acm_a_->NumberOfCodecs();
+  CodecInst my_codec_param;
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+    EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
+  }
+
+  // Test that unregister all receive codecs works.
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+    EXPECT_EQ(0, acm_b_->UnregisterReceiveCodec(my_codec_param.pltype));
+  }
+
+  // Register all available codes as receiving codecs once more.
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param));
+    EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param));
+  }
+
+  // Create and connect the channel.
+  channel_a2b_ = new TestPackStereo;
+  EXPECT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_));
+  channel_a2b_->RegisterReceiverACM(acm_b_.get());
+
+  // Start with setting VAD/DTX, before we know we will send stereo.
+  // Continue with setting a stereo codec as send codec and verify that
+  // VAD/DTX gets turned off.
+  EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
+  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
+  EXPECT_TRUE(dtx);
+  EXPECT_TRUE(vad);
+  char codec_pcma_temp[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma_temp, 8000, 64000, 80, 2, pcma_pltype_);
+  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
+  EXPECT_FALSE(dtx);
+  EXPECT_FALSE(vad);
+  if (test_mode_ != 0) {
+    printf("\n");
+  }
+
+  //
+  // Test Stereo-To-Stereo for all codecs.
+  //
+  audio_channels = 2;
+  codec_channels = 2;
+
+  // All codecs are tested for all allowed sampling frequencies, rates and
+  // packet sizes.
+#ifdef WEBRTC_CODEC_G722
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  channel_a2b_->set_codec_mode(kStereo);
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_g722[] = "G722";
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 320, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 480, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 640, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 800, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 960, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  channel_a2b_->set_codec_mode(kStereo);
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_l16[] = "L16";
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 160, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 240, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 320, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 320, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 480, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 640, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
+      l16_32khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 640, codec_channels,
+      l16_32khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  codec_channels = 2;
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_pcma[] = "PCMA";
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 160, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 240, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 320, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 400, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 480, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Test that VAD/DTX cannot be turned on while sending stereo.
+  EXPECT_EQ(-1, acm_a_->SetVAD(true, true, VADNormal));
+  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
+  EXPECT_FALSE(dtx);
+  EXPECT_FALSE(vad);
+  EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
+  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
+  EXPECT_FALSE(dtx);
+  EXPECT_FALSE(vad);
+
+  out_file_.Close();
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  char codec_pcmu[] = "PCMU";
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 160, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 240, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 320, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 400, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 480, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  if (test_mode_ != 0) {
+    printf("===========================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-stereo\n");
+  }
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  codec_channels = 2;
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+
+  char codec_opus[] = "opus";
+  // Run Opus with 10 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 20 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*2, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 40 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 60 ms frame size.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*6, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  // Run Opus with 20 ms frame size and different bitrates.
+  RegisterSendCodec('A', codec_opus, 48000, 40000, 960, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_opus, 48000, 510000, 960, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+  //
+  // Test Mono-To-Stereo for all codecs.
+  //
+  audio_channels = 1;
+  codec_channels = 2;
+
+#ifdef WEBRTC_CODEC_G722
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
+      g722_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
+      l16_32khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Mono-to-stereo\n");
+  }
+
+  // Keep encode and decode in stereo.
+  test_cntr_++;
+  channel_a2b_->set_codec_mode(kStereo);
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels,
+      opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in mono, decode in stereo mode.
+  RegisterSendCodec('A', codec_opus, 48000, 64000, 960, 1, opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+
+  //
+  // Test Stereo-To-Mono for all codecs.
+  //
+  audio_channels = 2;
+  codec_channels = 1;
+  channel_a2b_->set_codec_mode(kMono);
+
+#ifdef WEBRTC_CODEC_G722
+  // Run stereo audio and mono codec.
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
+      g722_pltype_);
+
+  // Make sure it is possible to set VAD/CNG, now that we are sending mono
+  // again.
+  EXPECT_EQ(0, acm_a_->SetVAD(true, true, VADNormal));
+  EXPECT_EQ(0, acm_a_->VAD(&dtx, &vad, &vad_mode));
+  EXPECT_TRUE(dtx);
+  EXPECT_TRUE(vad);
+  EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
+      l16_8khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
+      l16_16khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  if (test_mode_ != 0) {
+    printf("==============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
+      l16_32khz_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#ifdef PCMA_AND_PCMU
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  RegisterSendCodec('A', codec_pcmu, 8000, 64000, 80, codec_channels,
+                    pcmu_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  RegisterSendCodec('A', codec_pcma, 8000, 64000, 80, codec_channels,
+                    pcma_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  if (test_mode_ != 0) {
+    printf("===============================================================\n");
+    printf("Test number: %d\n", test_cntr_ + 1);
+    printf("Test type: Stereo-to-mono\n");
+  }
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  // Encode and decode in mono.
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels,
+      opus_pltype_);
+  CodecInst opus_codec_param;
+  for (uint8_t n = 0; n < num_encoders; n++) {
+    EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
+    if (!strcmp(opus_codec_param.plname, "opus")) {
+      opus_codec_param.channels = 1;
+      EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+      break;
+    }
+  }
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  // Encode in stereo, decode in mono.
+  RegisterSendCodec('A', codec_opus, 48000, 32000, 960, 2, opus_pltype_);
+  Run(channel_a2b_, audio_channels, codec_channels);
+
+  out_file_.Close();
+
+  // Test switching between decoding mono and stereo for Opus.
+
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: mono\n", test_cntr_);
+  }
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+  // Decode in stereo.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: stereo\n", test_cntr_);
+  }
+  opus_codec_param.channels = 2;
+  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  Run(channel_a2b_, audio_channels, 2);
+  out_file_.Close();
+  // Decode in mono.
+  test_cntr_++;
+  OpenOutFile(test_cntr_);
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
+        " Decode: mono\n", test_cntr_);
+  }
+  opus_codec_param.channels = 1;
+  EXPECT_EQ(0, acm_b_->RegisterReceiveCodec(opus_codec_param));
+  Run(channel_a2b_, audio_channels, codec_channels);
+  out_file_.Close();
+
+#endif
+
+  // Print out which codecs were tested, and which were not, in the run.
+  if (test_mode_ != 0) {
+    printf("\nThe following codecs was INCLUDED in the test:\n");
+#ifdef WEBRTC_CODEC_G722
+    printf("   G.722\n");
+#endif
+    printf("   PCM16\n");
+    printf("   G.711\n");
+#ifdef WEBRTC_CODEC_OPUS
+    printf("   Opus\n");
+#endif
+    printf("\nTo complete the test, listen to the %d number of output "
+           "files.\n",
+           test_cntr_);
+  }
+
+  // Delete the file pointers.
+  delete in_file_stereo_;
+  delete in_file_mono_;
+}
+
+// Register Codec to use in the test
+//
+// Input:   side             - which ACM to use, 'A' or 'B'
+//          codec_name       - name to use when register the codec
+//          sampling_freq_hz - sampling frequency in Herz
+//          rate             - bitrate in bytes
+//          pack_size        - packet size in samples
+//          channels         - number of channels; 1 for mono, 2 for stereo
+//          payload_type     - payload type for the codec
+void TestStereo::RegisterSendCodec(char side, char* codec_name,
+                                   int32_t sampling_freq_hz, int rate,
+                                   int pack_size, int channels,
+                                   int payload_type) {
+  if (test_mode_ != 0) {
+    // Print out codec and settings
+    printf("Codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
+           sampling_freq_hz, rate, pack_size);
+  }
+
+  // Store packet size in samples, used to validate the received packet
+  pack_size_samp_ = pack_size;
+
+  // Store the expected packet size in bytes, used to validate the received
+  // packet. Add 0.875 to always round up to a whole byte.
+  pack_size_bytes_ = (uint16_t)(static_cast<float>(pack_size * rate) /
+                                    static_cast<float>(sampling_freq_hz * 8) +
+                                0.875);
+
+  // Set pointer to the ACM where to register the codec
+  AudioCodingModule* my_acm = NULL;
+  switch (side) {
+    case 'A': {
+      my_acm = acm_a_.get();
+      break;
+    }
+    case 'B': {
+      my_acm = acm_b_.get();
+      break;
+    }
+    default:
+      break;
+  }
+  ASSERT_TRUE(my_acm != NULL);
+
+  CodecInst my_codec_param;
+  // Get all codec parameters before registering
+  EXPECT_GT(AudioCodingModule::Codec(codec_name, &my_codec_param,
+                                     sampling_freq_hz, channels), -1);
+  my_codec_param.rate = rate;
+  my_codec_param.pacsize = pack_size;
+  EXPECT_EQ(0, my_acm->RegisterSendCodec(my_codec_param));
+
+  send_codec_name_ = codec_name;
+}
+
+void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
+                     int percent_loss) {
+  AudioFrame audio_frame;
+
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  uint16_t rec_size;
+  uint32_t time_stamp_diff;
+  channel->reset_payload_size();
+  int error_count = 0;
+  int variable_bytes = 0;
+  int variable_packets = 0;
+
+  while (1) {
+    // Simulate packet loss by setting |packet_loss_| to "true" in
+    // |percent_loss| percent of the loops.
+    if (percent_loss > 0) {
+      if (counter_ == floor((100 / percent_loss) + 0.5)) {
+        counter_ = 0;
+        channel->set_lost_packet(true);
+      } else {
+        channel->set_lost_packet(false);
+      }
+      counter_++;
+    }
+
+    // Add 10 msec to ACM
+    if (in_channels == 1) {
+      if (in_file_mono_->EndOfFile()) {
+        break;
+      }
+      in_file_mono_->Read10MsData(audio_frame);
+    } else {
+      if (in_file_stereo_->EndOfFile()) {
+        break;
+      }
+      in_file_stereo_->Read10MsData(audio_frame);
+    }
+    EXPECT_GE(acm_a_->Add10MsData(audio_frame), 0);
+
+    // Verify that the received packet size matches the settings.
+    rec_size = channel->payload_size();
+    if ((0 < rec_size) & (rec_size < 65535)) {
+      if (strcmp(send_codec_name_, "opus") == 0) {
+        // Opus is a variable rate codec, hence calculate the average packet
+        // size, and later make sure the average is in the right range.
+        variable_bytes += rec_size;
+        variable_packets++;
+      } else {
+        // For fixed rate codecs, check that packet size is correct.
+        if ((rec_size != pack_size_bytes_ * out_channels)
+            && (pack_size_bytes_ < 65535)) {
+          error_count++;
+        }
+      }
+      // Verify that the timestamp is updated with expected length
+      time_stamp_diff = channel->timestamp_diff();
+      if ((counter_ > 10) && (time_stamp_diff != pack_size_samp_)) {
+        error_count++;
+      }
+    }
+
+    // Run received side of ACM
+    EXPECT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+
+    // Write output speech to file
+    out_file_.Write10MsData(
+        audio_frame.data_,
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+  }
+
+  EXPECT_EQ(0, error_count);
+
+  // Check that packet size is in the right range for variable rate codecs,
+  // such as Opus.
+  if (variable_packets > 0) {
+    variable_bytes /= variable_packets;
+    EXPECT_NEAR(variable_bytes, pack_size_bytes_, 3);
+  }
+
+  if (in_file_mono_->EndOfFile()) {
+    in_file_mono_->Rewind();
+  }
+  if (in_file_stereo_->EndOfFile()) {
+    in_file_stereo_->Rewind();
+  }
+  // Reset in case we ended with a lost packet
+  channel->set_lost_packet(false);
+}
+
+void TestStereo::OpenOutFile(int16_t test_number) {
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
+      << ".pcm";
+  file_name = file_stream.str();
+  out_file_.Open(file_name, 32000, "wb");
+}
+
+void TestStereo::DisplaySendReceiveCodec() {
+  auto send_codec = acm_a_->SendCodec();
+  if (test_mode_ != 0) {
+    ASSERT_TRUE(send_codec);
+    printf("%s -> ", send_codec->plname);
+  }
+  CodecInst receive_codec;
+  acm_b_->ReceiveCodec(&receive_codec);
+  if (test_mode_ != 0) {
+    printf("%s\n", receive_codec.plname);
+  }
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/TestStereo.h

@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+
+#include <math.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+
+#define PCMA_AND_PCMU
+
+namespace webrtc {
+
+enum StereoMonoMode {
+  kNotSet,
+  kMono,
+  kStereo
+};
+
+class TestPackStereo : public AudioPacketizationCallback {
+ public:
+  TestPackStereo();
+  ~TestPackStereo();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(const FrameType frame_type,
+                   const uint8_t payload_type,
+                   const uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   const size_t payload_size,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  uint16_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+  void set_codec_mode(StereoMonoMode mode);
+  void set_lost_packet(bool lost);
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  int16_t seq_no_;
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  int payload_size_;
+  StereoMonoMode codec_mode_;
+  // Simulate packet losses
+  bool lost_packet_;
+};
+
+class TestStereo : public ACMTest {
+ public:
+  explicit TestStereo(int test_mode);
+  ~TestStereo();
+
+  void Perform() override;
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name and a sampling frequncy matching is not required. This is useful
+  // for codecs which support several sampling frequency.
+  void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
+                         int rate, int pack_size, int channels,
+                         int payload_type);
+
+  void Run(TestPackStereo* channel, int in_channels, int out_channels,
+           int percent_loss = 0);
+  void OpenOutFile(int16_t test_number);
+  void DisplaySendReceiveCodec();
+
+  int test_mode_;
+
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+
+  TestPackStereo* channel_a2b_;
+
+  PCMFile* in_file_stereo_;
+  PCMFile* in_file_mono_;
+  PCMFile out_file_;
+  int16_t test_cntr_;
+  uint16_t pack_size_samp_;
+  uint16_t pack_size_bytes_;
+  int counter_;
+  char* send_codec_name_;
+
+  // Payload types for stereo codecs and CNG
+#ifdef WEBRTC_CODEC_G722
+  int g722_pltype_;
+#endif
+  int l16_8khz_pltype_;
+  int l16_16khz_pltype_;
+  int l16_32khz_pltype_;
+#ifdef PCMA_AND_PCMU
+  int pcma_pltype_;
+  int pcmu_pltype_;
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  int opus_pltype_;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_

webrtc/modules/audio_coding/test/TestVADDTX.cc

@@ -0,0 +1,271 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+
+#include <string>
+
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#ifdef WEBRTC_CODEC_ISAC
+const CodecInst kIsacWb = {103, "ISAC", 16000, 480, 1, 32000};
+const CodecInst kIsacSwb = {104, "ISAC", 32000, 960, 1, 56000};
+#endif
+
+#ifdef WEBRTC_CODEC_ILBC
+const CodecInst kIlbc = {102, "ILBC", 8000, 240, 1, 13300};
+#endif
+
+#ifdef WEBRTC_CODEC_OPUS
+const CodecInst kOpus = {120, "opus", 48000, 960, 1, 64000};
+const CodecInst kOpusStereo = {120, "opus", 48000, 960, 2, 64000};
+#endif
+
+ActivityMonitor::ActivityMonitor() {
+  ResetStatistics();
+}
+
+int32_t ActivityMonitor::InFrameType(FrameType frame_type) {
+  counter_[frame_type]++;
+  return 0;
+}
+
+void ActivityMonitor::PrintStatistics() {
+  printf("\n");
+  printf("kEmptyFrame       %u\n", counter_[kEmptyFrame]);
+  printf("kAudioFrameSpeech %u\n", counter_[kAudioFrameSpeech]);
+  printf("kAudioFrameCN     %u\n", counter_[kAudioFrameCN]);
+  printf("kVideoFrameKey    %u\n", counter_[kVideoFrameKey]);
+  printf("kVideoFrameDelta  %u\n", counter_[kVideoFrameDelta]);
+  printf("\n\n");
+}
+
+void ActivityMonitor::ResetStatistics() {
+  memset(counter_, 0, sizeof(counter_));
+}
+
+void ActivityMonitor::GetStatistics(uint32_t* counter) {
+  memcpy(counter, counter_, sizeof(counter_));
+}
+
+TestVadDtx::TestVadDtx()
+    : acm_send_(AudioCodingModule::Create(0)),
+      acm_receive_(AudioCodingModule::Create(1)),
+      channel_(new Channel),
+      monitor_(new ActivityMonitor) {
+  EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
+  channel_->RegisterReceiverACM(acm_receive_.get());
+  EXPECT_EQ(0, acm_send_->RegisterVADCallback(monitor_.get()));
+}
+
+void TestVadDtx::RegisterCodec(CodecInst codec_param) {
+  // Set the codec for sending and receiving.
+  EXPECT_EQ(0, acm_send_->RegisterSendCodec(codec_param));
+  EXPECT_EQ(0, acm_receive_->RegisterReceiveCodec(codec_param));
+  channel_->SetIsStereo(codec_param.channels > 1);
+}
+
+// Encoding a file and see if the numbers that various packets occur follow
+// the expectation.
+void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
+                     std::string out_filename, bool append,
+                     const int* expects) {
+  monitor_->ResetStatistics();
+
+  PCMFile in_file;
+  in_file.Open(in_filename, frequency, "rb");
+  in_file.ReadStereo(channels > 1);
+
+  PCMFile out_file;
+  if (append) {
+    out_file.Open(out_filename, kOutputFreqHz, "ab");
+  } else {
+    out_file.Open(out_filename, kOutputFreqHz, "wb");
+  }
+
+  uint16_t frame_size_samples = in_file.PayloadLength10Ms();
+  uint32_t time_stamp = 0x12345678;
+  AudioFrame audio_frame;
+  while (!in_file.EndOfFile()) {
+    in_file.Read10MsData(audio_frame);
+    audio_frame.timestamp_ = time_stamp;
+    time_stamp += frame_size_samples;
+    EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
+    acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame);
+    out_file.Write10MsData(audio_frame);
+  }
+
+  in_file.Close();
+  out_file.Close();
+
+#ifdef PRINT_STAT
+  monitor_->PrintStatistics();
+#endif
+
+  uint32_t stats[5];
+  monitor_->GetStatistics(stats);
+  monitor_->ResetStatistics();
+
+  for (const auto& st : stats) {
+    int i = &st - stats;  // Calculate the current position in stats.
+    switch (expects[i]) {
+      case 0: {
+        EXPECT_EQ(0u, st) << "stats[" << i << "] error.";
+        break;
+      }
+      case 1: {
+        EXPECT_GT(st, 0u) << "stats[" << i << "] error.";
+        break;
+      }
+    }
+  }
+}
+
+// Following is the implementation of TestWebRtcVadDtx.
+TestWebRtcVadDtx::TestWebRtcVadDtx()
+    : vad_enabled_(false),
+      dtx_enabled_(false),
+      output_file_num_(0) {
+}
+
+void TestWebRtcVadDtx::Perform() {
+  // Go through various test cases.
+#ifdef WEBRTC_CODEC_ISAC
+  // Register iSAC WB as send codec
+  RegisterCodec(kIsacWb);
+  RunTestCases();
+
+  // Register iSAC SWB as send codec
+  RegisterCodec(kIsacSwb);
+  RunTestCases();
+#endif
+
+#ifdef WEBRTC_CODEC_ILBC
+  // Register iLBC as send codec
+  RegisterCodec(kIlbc);
+  RunTestCases();
+#endif
+
+#ifdef WEBRTC_CODEC_OPUS
+  // Register Opus as send codec
+  RegisterCodec(kOpus);
+  RunTestCases();
+#endif
+}
+
+// Test various configurations on VAD/DTX.
+void TestWebRtcVadDtx::RunTestCases() {
+  // #1 DTX = OFF, VAD = OFF, VADNormal
+  SetVAD(false, false, VADNormal);
+  Test(true);
+
+  // #2 DTX = ON, VAD = ON, VADAggr
+  SetVAD(true, true, VADAggr);
+  Test(false);
+
+  // #3 DTX = ON, VAD = ON, VADLowBitrate
+  SetVAD(true, true, VADLowBitrate);
+  Test(false);
+
+  // #4 DTX = ON, VAD = ON, VADVeryAggr
+  SetVAD(true, true, VADVeryAggr);
+  Test(false);
+
+  // #5 DTX = ON, VAD = ON, VADNormal
+  SetVAD(true, true, VADNormal);
+  Test(false);
+}
+
+// Set the expectation and run the test.
+void TestWebRtcVadDtx::Test(bool new_outfile) {
+  int expects[] = {-1, 1, dtx_enabled_, 0, 0};
+  if (new_outfile) {
+    output_file_num_++;
+  }
+  std::stringstream out_filename;
+  out_filename << webrtc::test::OutputPath()
+               << "testWebRtcVadDtx_outFile_"
+               << output_file_num_
+               << ".pcm";
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+      32000, 1, out_filename.str(), !new_outfile, expects);
+}
+
+void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
+                              ACMVADMode vad_mode) {
+  ACMVADMode mode;
+  EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
+  EXPECT_EQ(0, acm_send_->VAD(&dtx_enabled_, &vad_enabled_, &mode));
+
+  auto codec_param = acm_send_->SendCodec();
+  ASSERT_TRUE(codec_param);
+  if (STR_CASE_CMP(codec_param->plname, "opus") == 0) {
+    // If send codec is Opus, WebRTC VAD/DTX cannot be used.
+    enable_dtx = enable_vad = false;
+  }
+
+  EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
+
+  if (dtx_enabled_) {
+    EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
+  } else {
+    // Using no DTX should not affect setting of VAD.
+    EXPECT_EQ(enable_vad, vad_enabled_);
+  }
+}
+
+// Following is the implementation of TestOpusDtx.
+void TestOpusDtx::Perform() {
+#ifdef WEBRTC_CODEC_ISAC
+  // If we set other codec than Opus, DTX cannot be toggled.
+  RegisterCodec(kIsacWb);
+  EXPECT_EQ(-1, acm_send_->EnableOpusDtx());
+  EXPECT_EQ(-1, acm_send_->DisableOpusDtx());
+#endif
+
+#ifdef WEBRTC_CODEC_OPUS
+  int expects[] = {0, 1, 0, 0, 0};
+
+  // Register Opus as send codec
+  std::string out_filename = webrtc::test::OutputPath() +
+      "testOpusDtx_outFile_mono.pcm";
+  RegisterCodec(kOpus);
+  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
+
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+      32000, 1, out_filename, false, expects);
+
+  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
+  expects[kEmptyFrame] = 1;
+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+      32000, 1, out_filename, true, expects);
+
+  // Register stereo Opus as send codec
+  out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
+  RegisterCodec(kOpusStereo);
+  EXPECT_EQ(0, acm_send_->DisableOpusDtx());
+  expects[kEmptyFrame] = 0;
+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
+      32000, 2, out_filename, false, expects);
+
+  EXPECT_EQ(0, acm_send_->EnableOpusDtx());
+
+  expects[kEmptyFrame] = 1;
+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
+      32000, 2, out_filename, true, expects);
+#endif
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/TestVADDTX.h

@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+
+namespace webrtc {
+
+class ActivityMonitor : public ACMVADCallback {
+ public:
+  ActivityMonitor();
+  int32_t InFrameType(FrameType frame_type);
+  void PrintStatistics();
+  void ResetStatistics();
+  void GetStatistics(uint32_t* stats);
+ private:
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  // 3 - kVideoFrameKey (not used by audio)
+  // 4 - kVideoFrameDelta (not used by audio)
+  uint32_t counter_[5];
+};
+
+
+// TestVadDtx is to verify that VAD/DTX perform as they should. It runs through
+// an audio file and check if the occurrence of various packet types follows
+// expectation. TestVadDtx needs its derived class to implement the Perform()
+// to put the test together.
+class TestVadDtx : public ACMTest {
+ public:
+  static const int kOutputFreqHz = 16000;
+
+  TestVadDtx();
+
+  virtual void Perform() = 0;
+
+ protected:
+  void RegisterCodec(CodecInst codec_param);
+
+  // Encoding a file and see if the numbers that various packets occur follow
+  // the expectation. Saves result to a file.
+  // expects[x] means
+  // -1 : do not care,
+  // 0  : there have been no packets of type |x|,
+  // 1  : there have been packets of type |x|,
+  // with |x| indicates the following packet types
+  // 0 - kEmptyFrame
+  // 1 - kAudioFrameSpeech
+  // 2 - kAudioFrameCN
+  // 3 - kVideoFrameKey (not used by audio)
+  // 4 - kVideoFrameDelta (not used by audio)
+  void Run(std::string in_filename, int frequency, int channels,
+           std::string out_filename, bool append, const int* expects);
+
+  rtc::scoped_ptr<AudioCodingModule> acm_send_;
+  rtc::scoped_ptr<AudioCodingModule> acm_receive_;
+  rtc::scoped_ptr<Channel> channel_;
+  rtc::scoped_ptr<ActivityMonitor> monitor_;
+};
+
+// TestWebRtcVadDtx is to verify that the WebRTC VAD/DTX perform as they should.
+class TestWebRtcVadDtx final : public TestVadDtx {
+ public:
+  TestWebRtcVadDtx();
+
+  void Perform() override;
+
+ private:
+  void RunTestCases();
+  void Test(bool new_outfile);
+  void SetVAD(bool enable_dtx, bool enable_vad, ACMVADMode vad_mode);
+
+  bool vad_enabled_;
+  bool dtx_enabled_;
+  int output_file_num_;
+};
+
+// TestOpusDtx is to verify that the Opus DTX performs as it should.
+class TestOpusDtx final : public TestVadDtx {
+ public:
+  void Perform() override;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_

webrtc/modules/audio_coding/test/Tester.cc

@@ -0,0 +1,171 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+using webrtc::Trace;
+
+// This parameter is used to describe how to run the tests. It is normally
+// set to 0, and all tests are run in quite mode.
+#define ACM_TEST_MODE 0
+
+TEST(AudioCodingModuleTest, TestAllCodecs) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+          "acm_allcodecs_trace.txt").c_str());
+  webrtc::TestAllCodecs(ACM_TEST_MODE).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestEncodeDecode)) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_encodedecode_trace.txt").c_str());
+  webrtc::EncodeDecodeTest(ACM_TEST_MODE).Perform();
+  Trace::ReturnTrace();
+}
+
+#ifdef WEBRTC_CODEC_RED
+#define IF_RED(x) x
+#else
+#define IF_RED(x) DISABLED_##x
+#endif
+
+TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_RED(TestRedFec))) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_fec_trace.txt").c_str());
+  webrtc::TestRedFec().Perform();
+  Trace::ReturnTrace();
+}
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
+TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(IF_ISAC(TestIsac))) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_isac_trace.txt").c_str());
+  webrtc::ISACTest(ACM_TEST_MODE).Perform();
+  Trace::ReturnTrace();
+}
+
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#define IF_ALL_CODECS(x) x
+#else
+#define IF_ALL_CODECS(x) DISABLED_##x
+#endif
+
+TEST(AudioCodingModuleTest,
+     DISABLED_ON_ANDROID(IF_ALL_CODECS(TwoWayCommunication))) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_twowaycom_trace.txt").c_str());
+  webrtc::TwoWayCommunication(ACM_TEST_MODE).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestStereo)) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_stereo_trace.txt").c_str());
+  webrtc::TestStereo(ACM_TEST_MODE).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, DISABLED_ON_ANDROID(TestWebRtcVadDtx)) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_vaddtx_trace.txt").c_str());
+  webrtc::TestWebRtcVadDtx().Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestOpusDtx) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_opusdtx_trace.txt").c_str());
+  webrtc::TestOpusDtx().Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestOpus) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_opus_trace.txt").c_str());
+  webrtc::OpusTest().Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLoss) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_packetloss_trace.txt").c_str());
+  webrtc::PacketLossTest(1, 10, 10, 1).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLossBurst) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_packetloss_burst_trace.txt").c_str());
+  webrtc::PacketLossTest(1, 10, 10, 2).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLossStereo) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_packetloss_trace.txt").c_str());
+  webrtc::PacketLossTest(2, 10, 10, 1).Perform();
+  Trace::ReturnTrace();
+}
+
+TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
+  Trace::CreateTrace();
+  Trace::SetTraceFile((webrtc::test::OutputPath() +
+      "acm_packetloss_burst_trace.txt").c_str());
+  webrtc::PacketLossTest(2, 10, 10, 2).Perform();
+  Trace::ReturnTrace();
+}
+
+// The full API test is too long to run automatically on bots, but can be used
+// for offline testing. User interaction is needed.
+#ifdef ACM_TEST_FULL_API
+  TEST(AudioCodingModuleTest, TestAPI) {
+    Trace::CreateTrace();
+    Trace::SetTraceFile((webrtc::test::OutputPath() +
+        "acm_apitest_trace.txt").c_str());
+    webrtc::APITest().Perform();
+    Trace::ReturnTrace();
+  }
+#endif

webrtc/modules/audio_coding/test/TimedTrace.cc

@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TimedTrace.h"
+#include <math.h>
+
+double TimedTrace::_timeEllapsedSec = 0;
+FILE* TimedTrace::_timedTraceFile = NULL;
+
+TimedTrace::TimedTrace() {
+
+}
+
+TimedTrace::~TimedTrace() {
+  if (_timedTraceFile != NULL) {
+    fclose(_timedTraceFile);
+  }
+  _timedTraceFile = NULL;
+}
+
+int16_t TimedTrace::SetUp(char* fileName) {
+  if (_timedTraceFile == NULL) {
+    _timedTraceFile = fopen(fileName, "w");
+  }
+  if (_timedTraceFile == NULL) {
+    return -1;
+  }
+  return 0;
+}
+
+void TimedTrace::SetTimeEllapsed(double timeEllapsedSec) {
+  _timeEllapsedSec = timeEllapsedSec;
+}
+
+double TimedTrace::TimeEllapsed() {
+  return _timeEllapsedSec;
+}
+
+void TimedTrace::Tick10Msec() {
+  _timeEllapsedSec += 0.010;
+}
+
+void TimedTrace::TimedLogg(char* message) {
+  unsigned int minutes = (uint32_t) floor(_timeEllapsedSec / 60.0);
+  double seconds = _timeEllapsedSec - minutes * 60;
+  //char myFormat[100] = "%8.2f, %3u:%05.2f: %s\n";
+  if (_timedTraceFile != NULL) {
+    fprintf(_timedTraceFile, "%8.2f, %3u:%05.2f: %s\n", _timeEllapsedSec,
+            minutes, seconds, message);
+  }
+}

webrtc/modules/audio_coding/test/TimedTrace.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_
+
+#include "webrtc/typedefs.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+
+class TimedTrace {
+ public:
+  TimedTrace();
+  ~TimedTrace();
+
+  void SetTimeEllapsed(double myTime);
+  double TimeEllapsed();
+  void Tick10Msec();
+  int16_t SetUp(char* fileName);
+  void TimedLogg(char* message);
+
+ private:
+  static double _timeEllapsedSec;
+  static FILE* _timedTraceFile;
+
+};
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_

webrtc/modules/audio_coding/test/TwoWayCommunication.cc

@@ -0,0 +1,299 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "TwoWayCommunication.h"
+
+#include <ctype.h>
+#include <stdio.h>
+#include <string.h>
+
+#ifdef WIN32
+#include <Windows.h>
+#endif
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+
+TwoWayCommunication::TwoWayCommunication(int testMode)
+    : _acmA(AudioCodingModule::Create(1)),
+      _acmRefA(AudioCodingModule::Create(3)),
+      _testMode(testMode) {
+  AudioCodingModule::Config config;
+  // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
+  config.neteq_config.playout_mode = kPlayoutFax;
+  config.id = 2;
+  _acmB.reset(AudioCodingModule::Create(config));
+  config.id = 4;
+  _acmRefB.reset(AudioCodingModule::Create(config));
+}
+
+TwoWayCommunication::~TwoWayCommunication() {
+  delete _channel_A2B;
+  delete _channel_B2A;
+  delete _channelRef_A2B;
+  delete _channelRef_B2A;
+#ifdef WEBRTC_DTMF_DETECTION
+  if (_dtmfDetectorA != NULL) {
+    delete _dtmfDetectorA;
+  }
+  if (_dtmfDetectorB != NULL) {
+    delete _dtmfDetectorB;
+  }
+#endif
+  _inFileA.Close();
+  _inFileB.Close();
+  _outFileA.Close();
+  _outFileB.Close();
+  _outFileRefA.Close();
+  _outFileRefB.Close();
+}
+
+void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
+                                      uint8_t* codecID_B) {
+  rtc::scoped_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(0));
+  uint8_t noCodec = tmpACM->NumberOfCodecs();
+  CodecInst codecInst;
+  printf("List of Supported Codecs\n");
+  printf("========================\n");
+  for (uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++) {
+    EXPECT_EQ(tmpACM->Codec(codecCntr, &codecInst), 0);
+    printf("%d- %s\n", codecCntr, codecInst.plname);
+  }
+  printf("\nChoose a send codec for side A [0]: ");
+  char myStr[15] = "";
+  EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+  *codecID_A = (uint8_t) atoi(myStr);
+
+  printf("\nChoose a send codec for side B [0]: ");
+  EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+  *codecID_B = (uint8_t) atoi(myStr);
+
+  printf("\n");
+}
+
+void TwoWayCommunication::SetUp() {
+  uint8_t codecID_A;
+  uint8_t codecID_B;
+
+  ChooseCodec(&codecID_A, &codecID_B);
+  CodecInst codecInst_A;
+  CodecInst codecInst_B;
+  CodecInst dummyCodec;
+  EXPECT_EQ(0, _acmA->Codec(codecID_A, &codecInst_A));
+  EXPECT_EQ(0, _acmB->Codec(codecID_B, &codecInst_B));
+  EXPECT_EQ(0, _acmA->Codec(6, &dummyCodec));
+
+  //--- Set A codecs
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
+  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+  //--- Set ref-A codecs
+  EXPECT_EQ(0, _acmRefA->RegisterSendCodec(codecInst_A));
+  EXPECT_EQ(0, _acmRefA->RegisterReceiveCodec(codecInst_B));
+
+  //--- Set B codecs
+  EXPECT_EQ(0, _acmB->RegisterSendCodec(codecInst_B));
+  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(codecInst_A));
+
+  //--- Set ref-B codecs
+  EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
+  EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
+
+  uint16_t frequencyHz;
+
+  //--- Input A
+  std::string in_file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  frequencyHz = 32000;
+  printf("Enter input file at side A [%s]: ", in_file_name.c_str());
+  PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
+  _inFileA.Open(in_file_name, frequencyHz, "rb");
+
+  //--- Output A
+  std::string out_file_a = webrtc::test::OutputPath() + "outA.pcm";
+  printf("Output file at side A: %s\n", out_file_a.c_str());
+  printf("Sampling frequency (in Hz) of the above file: %u\n", frequencyHz);
+  _outFileA.Open(out_file_a, frequencyHz, "wb");
+  std::string ref_file_name = webrtc::test::OutputPath() + "ref_outA.pcm";
+  _outFileRefA.Open(ref_file_name, frequencyHz, "wb");
+
+  //--- Input B
+  in_file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
+                                            "pcm");
+  frequencyHz = 32000;
+  printf("\n\nEnter input file at side B [%s]: ", in_file_name.c_str());
+  PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
+  _inFileB.Open(in_file_name, frequencyHz, "rb");
+
+  //--- Output B
+  std::string out_file_b = webrtc::test::OutputPath() + "outB.pcm";
+  printf("Output file at side B: %s\n", out_file_b.c_str());
+  printf("Sampling frequency (in Hz) of the above file: %u\n", frequencyHz);
+  _outFileB.Open(out_file_b, frequencyHz, "wb");
+  ref_file_name = webrtc::test::OutputPath() + "ref_outB.pcm";
+  _outFileRefB.Open(ref_file_name, frequencyHz, "wb");
+
+  //--- Set A-to-B channel
+  _channel_A2B = new Channel;
+  _acmA->RegisterTransportCallback(_channel_A2B);
+  _channel_A2B->RegisterReceiverACM(_acmB.get());
+  //--- Do the same for the reference
+  _channelRef_A2B = new Channel;
+  _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+  _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
+
+  //--- Set B-to-A channel
+  _channel_B2A = new Channel;
+  _acmB->RegisterTransportCallback(_channel_B2A);
+  _channel_B2A->RegisterReceiverACM(_acmA.get());
+  //--- Do the same for reference
+  _channelRef_B2A = new Channel;
+  _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+  _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
+}
+
+void TwoWayCommunication::SetUpAutotest() {
+  CodecInst codecInst_A;
+  CodecInst codecInst_B;
+  CodecInst dummyCodec;
+
+  EXPECT_EQ(0, _acmA->Codec("ISAC", &codecInst_A, 16000, 1));
+  EXPECT_EQ(0, _acmB->Codec("L16", &codecInst_B, 8000, 1));
+  EXPECT_EQ(0, _acmA->Codec(6, &dummyCodec));
+
+  //--- Set A codecs
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(codecInst_A));
+  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(codecInst_B));
+
+  //--- Set ref-A codecs
+  EXPECT_GT(_acmRefA->RegisterSendCodec(codecInst_A), -1);
+  EXPECT_GT(_acmRefA->RegisterReceiveCodec(codecInst_B), -1);
+
+  //--- Set B codecs
+  EXPECT_GT(_acmB->RegisterSendCodec(codecInst_B), -1);
+  EXPECT_GT(_acmB->RegisterReceiveCodec(codecInst_A), -1);
+
+  //--- Set ref-B codecs
+  EXPECT_EQ(0, _acmRefB->RegisterSendCodec(codecInst_B));
+  EXPECT_EQ(0, _acmRefB->RegisterReceiveCodec(codecInst_A));
+
+  uint16_t frequencyHz;
+
+  //--- Input A and B
+  std::string in_file_name = webrtc::test::ResourcePath(
+      "audio_coding/testfile32kHz", "pcm");
+  frequencyHz = 16000;
+  _inFileA.Open(in_file_name, frequencyHz, "rb");
+  _inFileB.Open(in_file_name, frequencyHz, "rb");
+
+  //--- Output A
+  std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
+  frequencyHz = 16000;
+  _outFileA.Open(output_file_a, frequencyHz, "wb");
+  std::string output_ref_file_a = webrtc::test::OutputPath()
+      + "ref_outAutotestA.pcm";
+  _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
+
+  //--- Output B
+  std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
+  frequencyHz = 16000;
+  _outFileB.Open(output_file_b, frequencyHz, "wb");
+  std::string output_ref_file_b = webrtc::test::OutputPath()
+      + "ref_outAutotestB.pcm";
+  _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
+
+  //--- Set A-to-B channel
+  _channel_A2B = new Channel;
+  _acmA->RegisterTransportCallback(_channel_A2B);
+  _channel_A2B->RegisterReceiverACM(_acmB.get());
+  //--- Do the same for the reference
+  _channelRef_A2B = new Channel;
+  _acmRefA->RegisterTransportCallback(_channelRef_A2B);
+  _channelRef_A2B->RegisterReceiverACM(_acmRefB.get());
+
+  //--- Set B-to-A channel
+  _channel_B2A = new Channel;
+  _acmB->RegisterTransportCallback(_channel_B2A);
+  _channel_B2A->RegisterReceiverACM(_acmA.get());
+  //--- Do the same for reference
+  _channelRef_B2A = new Channel;
+  _acmRefB->RegisterTransportCallback(_channelRef_B2A);
+  _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
+}
+
+void TwoWayCommunication::Perform() {
+  if (_testMode == 0) {
+    SetUpAutotest();
+  } else {
+    SetUp();
+  }
+  unsigned int msecPassed = 0;
+  unsigned int secPassed = 0;
+
+  int32_t outFreqHzA = _outFileA.SamplingFrequency();
+  int32_t outFreqHzB = _outFileB.SamplingFrequency();
+
+  AudioFrame audioFrame;
+
+  auto codecInst_B = _acmB->SendCodec();
+  ASSERT_TRUE(codecInst_B);
+
+  // In the following loop we tests that the code can handle misuse of the APIs.
+  // In the middle of a session with data flowing between two sides, called A
+  // and B, APIs will be called, and the code should continue to run, and be
+  // able to recover.
+  while (!_inFileA.EndOfFile() && !_inFileB.EndOfFile()) {
+    msecPassed += 10;
+    EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+    EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+    EXPECT_GE(_acmRefA->Add10MsData(audioFrame), 0);
+
+    EXPECT_GT(_inFileB.Read10MsData(audioFrame), 0);
+
+    EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
+    EXPECT_GE(_acmRefB->Add10MsData(audioFrame), 0);
+    EXPECT_EQ(0, _acmA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+    _outFileA.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmRefA->PlayoutData10Ms(outFreqHzA, &audioFrame));
+    _outFileRefA.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    _outFileB.Write10MsData(audioFrame);
+    EXPECT_EQ(0, _acmRefB->PlayoutData10Ms(outFreqHzB, &audioFrame));
+    _outFileRefB.Write10MsData(audioFrame);
+
+    // Update time counters each time a second of data has passed.
+    if (msecPassed >= 1000) {
+      msecPassed = 0;
+      secPassed++;
+    }
+    // Re-register send codec on side B.
+    if (((secPassed % 5) == 4) && (msecPassed >= 990)) {
+      EXPECT_EQ(0, _acmB->RegisterSendCodec(*codecInst_B));
+      EXPECT_TRUE(_acmB->SendCodec());
+    }
+    // Initialize receiver on side A.
+    if (((secPassed % 7) == 6) && (msecPassed == 0))
+      EXPECT_EQ(0, _acmA->InitializeReceiver());
+    // Re-register codec on side A.
+    if (((secPassed % 7) == 6) && (msecPassed >= 990)) {
+      EXPECT_EQ(0, _acmA->RegisterReceiveCodec(*codecInst_B));
+    }
+  }
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/TwoWayCommunication.h

@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+
+namespace webrtc {
+
+class TwoWayCommunication : public ACMTest {
+ public:
+  explicit TwoWayCommunication(int testMode);
+  ~TwoWayCommunication();
+
+  void Perform();
+ private:
+  void ChooseCodec(uint8_t* codecID_A, uint8_t* codecID_B);
+  void SetUp();
+  void SetUpAutotest();
+
+  rtc::scoped_ptr<AudioCodingModule> _acmA;
+  rtc::scoped_ptr<AudioCodingModule> _acmB;
+
+  rtc::scoped_ptr<AudioCodingModule> _acmRefA;
+  rtc::scoped_ptr<AudioCodingModule> _acmRefB;
+
+  Channel* _channel_A2B;
+  Channel* _channel_B2A;
+
+  Channel* _channelRef_A2B;
+  Channel* _channelRef_B2A;
+
+  PCMFile _inFileA;
+  PCMFile _inFileB;
+
+  PCMFile _outFileA;
+  PCMFile _outFileB;
+
+  PCMFile _outFileRefA;
+  PCMFile _outFileRefB;
+
+  int _testMode;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_

webrtc/modules/audio_coding/test/delay_test.cc

@@ -0,0 +1,265 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <assert.h>
+#include <math.h>
+
+#include <iostream>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+DEFINE_string(codec, "isac", "Codec Name");
+DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
+DEFINE_int32(num_channels, 1, "Number of Channels.");
+DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
+DEFINE_int32(delay, 0, "Delay in millisecond.");
+DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
+DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
+DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
+
+namespace webrtc {
+
+namespace {
+
+struct CodecSettings {
+  char name[50];
+  int sample_rate_hz;
+  int num_channels;
+};
+
+struct AcmSettings {
+  bool dtx;
+  bool fec;
+};
+
+struct TestSettings {
+  CodecSettings codec;
+  AcmSettings acm;
+  bool packet_loss;
+};
+
+}  // namespace
+
+class DelayTest {
+ public:
+  DelayTest()
+      : acm_a_(AudioCodingModule::Create(0)),
+        acm_b_(AudioCodingModule::Create(1)),
+        channel_a2b_(new Channel),
+        test_cntr_(0),
+        encoding_sample_rate_hz_(8000) {}
+
+  ~DelayTest() {
+    if (channel_a2b_ != NULL) {
+      delete channel_a2b_;
+      channel_a2b_ = NULL;
+    }
+    in_file_a_.Close();
+  }
+
+  void Initialize() {
+    test_cntr_ = 0;
+    std::string file_name = webrtc::test::ResourcePath(
+        "audio_coding/testfile32kHz", "pcm");
+    if (FLAGS_input_file.size() > 0)
+      file_name = FLAGS_input_file;
+    in_file_a_.Open(file_name, 32000, "rb");
+    ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
+        "Couldn't initialize receiver.\n";
+    ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
+        "Couldn't initialize receiver.\n";
+
+    if (FLAGS_delay > 0) {
+      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
+          "Failed to set minimum delay.\n";
+    }
+
+    int num_encoders = acm_a_->NumberOfCodecs();
+    CodecInst my_codec_param;
+    for (int n = 0; n < num_encoders; n++) {
+      EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
+          "Failed to get codec.";
+      if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
+        my_codec_param.channels = 1;
+      else if (my_codec_param.channels > 1)
+        continue;
+      if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
+          my_codec_param.plfreq == 48000)
+        continue;
+      if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
+        continue;
+      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
+          "Couldn't register receive codec.\n";
+    }
+
+    // Create and connect the channel
+    ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
+        "Couldn't register Transport callback.\n";
+    channel_a2b_->RegisterReceiverACM(acm_b_.get());
+  }
+
+  void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
+               const char* output_prefix) {
+    for (size_t n = 0; n < num_tests; ++n) {
+      ApplyConfig(config[n]);
+      Run(duration_sec, output_prefix);
+    }
+  }
+
+ private:
+  void ApplyConfig(const TestSettings& config) {
+    printf("====================================\n");
+    printf("Test %d \n"
+           "Codec: %s, %d kHz, %d channel(s)\n"
+           "ACM: DTX %s, FEC %s\n"
+           "Channel: %s\n",
+           ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
+           config.codec.num_channels, config.acm.dtx ? "on" : "off",
+           config.acm.fec ? "on" : "off",
+           config.packet_loss ? "with packet-loss" : "no packet-loss");
+    SendCodec(config.codec);
+    ConfigAcm(config.acm);
+    ConfigChannel(config.packet_loss);
+  }
+
+  void SendCodec(const CodecSettings& config) {
+    CodecInst my_codec_param;
+    ASSERT_EQ(0, AudioCodingModule::Codec(
+              config.name, &my_codec_param, config.sample_rate_hz,
+              config.num_channels)) << "Specified codec is not supported.\n";
+
+    encoding_sample_rate_hz_ = my_codec_param.plfreq;
+    ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
+        "Failed to register send-codec.\n";
+  }
+
+  void ConfigAcm(const AcmSettings& config) {
+    ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
+        "Failed to set VAD.\n";
+    ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
+        "Failed to set RED.\n";
+  }
+
+  void ConfigChannel(bool packet_loss) {
+    channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
+  }
+
+  void OpenOutFile(const char* output_id) {
+    std::stringstream file_stream;
+    file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
+        << "Hz" << "_" << FLAGS_delay << "ms.pcm";
+    std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
+    std::string file_name = webrtc::test::OutputPath() + file_stream.str();
+    out_file_b_.Open(file_name.c_str(), 32000, "wb");
+  }
+
+  void Run(int duration_sec, const char* output_prefix) {
+    OpenOutFile(output_prefix);
+    AudioFrame audio_frame;
+    uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
+
+    int num_frames = 0;
+    int in_file_frames = 0;
+    uint32_t playout_ts;
+    uint32_t received_ts;
+    double average_delay = 0;
+    double inst_delay_sec = 0;
+    while (num_frames < (duration_sec * 100)) {
+      if (in_file_a_.EndOfFile()) {
+        in_file_a_.Rewind();
+      }
+
+      // Print delay information every 16 frame
+      if ((num_frames & 0x3F) == 0x3F) {
+        NetworkStatistics statistics;
+        acm_b_->GetNetworkStatistics(&statistics);
+        fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
+                " ts-based average = %6.3f, "
+                "curr buff-lev = %4u opt buff-lev = %4u \n",
+                statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
+                statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
+                average_delay, statistics.currentBufferSize,
+                statistics.preferredBufferSize);
+        fflush (stdout);
+      }
+
+      in_file_a_.Read10MsData(audio_frame);
+      ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
+      ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+      out_file_b_.Write10MsData(
+          audio_frame.data_,
+          audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+      acm_b_->PlayoutTimestamp(&playout_ts);
+      received_ts = channel_a2b_->LastInTimestamp();
+      inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
+          / static_cast<double>(encoding_sample_rate_hz_);
+
+      if (num_frames > 10)
+        average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
+
+      ++num_frames;
+      ++in_file_frames;
+    }
+    out_file_b_.Close();
+  }
+
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+
+  Channel* channel_a2b_;
+
+  PCMFile in_file_a_;
+  PCMFile out_file_b_;
+  int test_cntr_;
+  int encoding_sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::TestSettings test_setting;
+  strcpy(test_setting.codec.name, FLAGS_codec.c_str());
+
+  if (FLAGS_sample_rate_hz != 8000 &&
+      FLAGS_sample_rate_hz != 16000 &&
+      FLAGS_sample_rate_hz != 32000 &&
+      FLAGS_sample_rate_hz != 48000) {
+    std::cout << "Invalid sampling rate.\n";
+    return 1;
+  }
+  test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
+  if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
+    std::cout << "Only mono and stereo are supported.\n";
+    return 1;
+  }
+  test_setting.codec.num_channels = FLAGS_num_channels;
+  test_setting.acm.dtx = FLAGS_dtx;
+  test_setting.acm.fec = FLAGS_fec;
+  test_setting.packet_loss = FLAGS_packet_loss;
+
+  webrtc::DelayTest delay_test;
+  delay_test.Initialize();
+  delay_test.Perform(&test_setting, 1, 240, "delay_test");
+  return 0;
+}

webrtc/modules/audio_coding/test/iSACTest.cc

@@ -0,0 +1,339 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/iSACTest.h"
+
+#include <ctype.h>
+#include <stdio.h>
+#include <string.h>
+
+#if _WIN32
+#include <windows.h>
+#elif WEBRTC_LINUX
+#include <time.h>
+#else
+#include <sys/time.h>
+#include <time.h>
+#endif
+
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/event_wrapper.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
+  isacConfig.currentRateBitPerSec = 0;
+  isacConfig.currentFrameSizeMsec = 0;
+  isacConfig.encodingMode = -1;
+  isacConfig.initRateBitPerSec = 0;
+  isacConfig.initFrameSizeInMsec = 0;
+  isacConfig.enforceFrameSize = false;
+  return;
+}
+
+int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
+                     int testMode) {
+
+  if ((isacConfig.currentRateBitPerSec != 0)
+      || (isacConfig.currentFrameSizeMsec != 0)) {
+    auto sendCodec = acm->SendCodec();
+    EXPECT_TRUE(sendCodec);
+    if (isacConfig.currentRateBitPerSec < 0) {
+      // Register iSAC in adaptive (channel-dependent) mode.
+      sendCodec->rate = -1;
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
+    } else {
+      if (isacConfig.currentRateBitPerSec != 0) {
+        sendCodec->rate = isacConfig.currentRateBitPerSec;
+      }
+      if (isacConfig.currentFrameSizeMsec != 0) {
+        sendCodec->pacsize = isacConfig.currentFrameSizeMsec
+            * (sendCodec->plfreq / 1000);
+      }
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
+    }
+  }
+
+  return 0;
+}
+
+ISACTest::ISACTest(int testMode)
+    : _acmA(AudioCodingModule::Create(1)),
+      _acmB(AudioCodingModule::Create(2)),
+      _testMode(testMode) {}
+
+ISACTest::~ISACTest() {}
+
+void ISACTest::Setup() {
+  int codecCntr;
+  CodecInst codecParam;
+
+  for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
+      codecCntr++) {
+    EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
+    if (!STR_CASE_CMP(codecParam.plname, "ISAC")
+        && codecParam.plfreq == 16000) {
+      memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
+      _idISAC16kHz = codecCntr;
+    }
+    if (!STR_CASE_CMP(codecParam.plname, "ISAC")
+        && codecParam.plfreq == 32000) {
+      memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
+      _idISAC32kHz = codecCntr;
+    }
+  }
+
+  // Register both iSAC-wb & iSAC-swb in both sides as receiver codecs.
+  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC16kHz));
+  EXPECT_EQ(0, _acmA->RegisterReceiveCodec(_paramISAC32kHz));
+  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC16kHz));
+  EXPECT_EQ(0, _acmB->RegisterReceiveCodec(_paramISAC32kHz));
+
+  //--- Set A-to-B channel
+  _channel_A2B.reset(new Channel);
+  EXPECT_EQ(0, _acmA->RegisterTransportCallback(_channel_A2B.get()));
+  _channel_A2B->RegisterReceiverACM(_acmB.get());
+
+  //--- Set B-to-A channel
+  _channel_B2A.reset(new Channel);
+  EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
+  _channel_B2A->RegisterReceiverACM(_acmA.get());
+
+  file_name_swb_ = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
+                                              "pcm");
+
+  EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
+
+  _inFileA.Open(file_name_swb_, 32000, "rb");
+  std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
+  std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
+  _outFileA.Open(fileNameA, 32000, "wb");
+  _outFileB.Open(fileNameB, 32000, "wb");
+
+  while (!_inFileA.EndOfFile()) {
+    Run10ms();
+  }
+  CodecInst receiveCodec;
+  EXPECT_EQ(0, _acmA->ReceiveCodec(&receiveCodec));
+  EXPECT_EQ(0, _acmB->ReceiveCodec(&receiveCodec));
+
+  _inFileA.Close();
+  _outFileA.Close();
+  _outFileB.Close();
+}
+
+void ISACTest::Perform() {
+  Setup();
+
+  int16_t testNr = 0;
+  ACMTestISACConfig wbISACConfig;
+  ACMTestISACConfig swbISACConfig;
+
+  SetISACConfigDefault(wbISACConfig);
+  SetISACConfigDefault(swbISACConfig);
+
+  wbISACConfig.currentRateBitPerSec = -1;
+  swbISACConfig.currentRateBitPerSec = -1;
+  testNr++;
+  EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+  if (_testMode != 0) {
+    SetISACConfigDefault(wbISACConfig);
+    SetISACConfigDefault(swbISACConfig);
+
+    wbISACConfig.currentRateBitPerSec = -1;
+    swbISACConfig.currentRateBitPerSec = -1;
+    wbISACConfig.initRateBitPerSec = 13000;
+    wbISACConfig.initFrameSizeInMsec = 60;
+    swbISACConfig.initRateBitPerSec = 20000;
+    swbISACConfig.initFrameSizeInMsec = 30;
+    testNr++;
+    EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+    SetISACConfigDefault(wbISACConfig);
+    SetISACConfigDefault(swbISACConfig);
+
+    wbISACConfig.currentRateBitPerSec = 20000;
+    swbISACConfig.currentRateBitPerSec = 48000;
+    testNr++;
+    EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+    wbISACConfig.currentRateBitPerSec = 16000;
+    swbISACConfig.currentRateBitPerSec = 30000;
+    wbISACConfig.currentFrameSizeMsec = 60;
+    testNr++;
+    EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+  }
+
+  SetISACConfigDefault(wbISACConfig);
+  SetISACConfigDefault(swbISACConfig);
+  testNr++;
+  EncodeDecode(testNr, wbISACConfig, swbISACConfig);
+
+  testNr++;
+  if (_testMode == 0) {
+    SwitchingSamplingRate(testNr, 4);
+  } else {
+    SwitchingSamplingRate(testNr, 80);
+  }
+}
+
+void ISACTest::Run10ms() {
+  AudioFrame audioFrame;
+  EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0);
+  EXPECT_GE(_acmA->Add10MsData(audioFrame), 0);
+  EXPECT_GE(_acmB->Add10MsData(audioFrame), 0);
+  EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame));
+  _outFileA.Write10MsData(audioFrame);
+  EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame));
+  _outFileB.Write10MsData(audioFrame);
+}
+
+void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
+                            ACMTestISACConfig& swbISACConfig) {
+  // Files in Side A and B
+  _inFileA.Open(file_name_swb_, 32000, "rb", true);
+  _inFileB.Open(file_name_swb_, 32000, "rb", true);
+
+  std::string file_name_out;
+  std::stringstream file_stream_a;
+  std::stringstream file_stream_b;
+  file_stream_a << webrtc::test::OutputPath();
+  file_stream_b << webrtc::test::OutputPath();
+  file_stream_a << "out_iSACTest_A_" << testNr << ".pcm";
+  file_stream_b << "out_iSACTest_B_" << testNr << ".pcm";
+  file_name_out = file_stream_a.str();
+  _outFileA.Open(file_name_out, 32000, "wb");
+  file_name_out = file_stream_b.str();
+  _outFileB.Open(file_name_out, 32000, "wb");
+
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC16kHz));
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
+  EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC32kHz));
+  EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
+
+  // Side A is sending super-wideband, and side B is sending wideband.
+  SetISAConfig(swbISACConfig, _acmA.get(), _testMode);
+  SetISAConfig(wbISACConfig, _acmB.get(), _testMode);
+
+  bool adaptiveMode = false;
+  if ((swbISACConfig.currentRateBitPerSec == -1)
+      || (wbISACConfig.currentRateBitPerSec == -1)) {
+    adaptiveMode = true;
+  }
+  _myTimer.Reset();
+  _channel_A2B->ResetStats();
+  _channel_B2A->ResetStats();
+
+  char currentTime[500];
+  EventTimerWrapper* myEvent = EventTimerWrapper::Create();
+  EXPECT_TRUE(myEvent->StartTimer(true, 10));
+  while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
+    Run10ms();
+    _myTimer.Tick10ms();
+    _myTimer.CurrentTimeHMS(currentTime);
+
+    if ((adaptiveMode) && (_testMode != 0)) {
+      myEvent->Wait(5000);
+      EXPECT_TRUE(_acmA->SendCodec());
+      EXPECT_TRUE(_acmB->SendCodec());
+    }
+  }
+
+  if (_testMode != 0) {
+    printf("\n\nSide A statistics\n\n");
+    _channel_A2B->PrintStats(_paramISAC32kHz);
+
+    printf("\n\nSide B statistics\n\n");
+    _channel_B2A->PrintStats(_paramISAC16kHz);
+  }
+
+  _channel_A2B->ResetStats();
+  _channel_B2A->ResetStats();
+
+  _outFileA.Close();
+  _outFileB.Close();
+  _inFileA.Close();
+  _inFileB.Close();
+}
+
+void ISACTest::SwitchingSamplingRate(int testNr, int maxSampRateChange) {
+  // Files in Side A
+  _inFileA.Open(file_name_swb_, 32000, "rb");
+  _inFileB.Open(file_name_swb_, 32000, "rb");
+
+  std::string file_name_out;
+  std::stringstream file_stream_a;
+  std::stringstream file_stream_b;
+  file_stream_a << webrtc::test::OutputPath();
+  file_stream_b << webrtc::test::OutputPath();
+  file_stream_a << "out_iSACTest_A_" << testNr << ".pcm";
+  file_stream_b << "out_iSACTest_B_" << testNr << ".pcm";
+  file_name_out = file_stream_a.str();
+  _outFileA.Open(file_name_out, 32000, "wb");
+  file_name_out = file_stream_b.str();
+  _outFileB.Open(file_name_out, 32000, "wb");
+
+  // Start with side A sending super-wideband and side B seding wideband.
+  // Toggle sending wideband/super-wideband in this test.
+  EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
+  EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
+
+  int numSendCodecChanged = 0;
+  _myTimer.Reset();
+  char currentTime[50];
+  while (numSendCodecChanged < (maxSampRateChange << 1)) {
+    Run10ms();
+    _myTimer.Tick10ms();
+    _myTimer.CurrentTimeHMS(currentTime);
+    if (_testMode == 2)
+      printf("\r%s", currentTime);
+    if (_inFileA.EndOfFile()) {
+      if (_inFileA.SamplingFrequency() == 16000) {
+        // Switch side A to send super-wideband.
+        _inFileA.Close();
+        _inFileA.Open(file_name_swb_, 32000, "rb");
+        EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
+      } else {
+        // Switch side A to send wideband.
+        _inFileA.Close();
+        _inFileA.Open(file_name_swb_, 32000, "rb");
+        EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC16kHz));
+      }
+      numSendCodecChanged++;
+    }
+
+    if (_inFileB.EndOfFile()) {
+      if (_inFileB.SamplingFrequency() == 16000) {
+        // Switch side B to send super-wideband.
+        _inFileB.Close();
+        _inFileB.Open(file_name_swb_, 32000, "rb");
+        EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC32kHz));
+      } else {
+        // Switch side B to send wideband.
+        _inFileB.Close();
+        _inFileB.Open(file_name_swb_, 32000, "rb");
+        EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
+      }
+      numSendCodecChanged++;
+    }
+  }
+  _outFileA.Close();
+  _outFileB.Close();
+  _inFileA.Close();
+  _inFileB.Close();
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/iSACTest.h

@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+
+#include <string.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+
+#define MAX_FILE_NAME_LENGTH_BYTE 500
+#define NO_OF_CLIENTS             15
+
+namespace webrtc {
+
+struct ACMTestISACConfig {
+  int32_t currentRateBitPerSec;
+  int16_t currentFrameSizeMsec;
+  int16_t encodingMode;
+  uint32_t initRateBitPerSec;
+  int16_t initFrameSizeInMsec;
+  bool enforceFrameSize;
+};
+
+class ISACTest : public ACMTest {
+ public:
+  explicit ISACTest(int testMode);
+  ~ISACTest();
+
+  void Perform();
+ private:
+  void Setup();
+
+  void Run10ms();
+
+  void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
+                    ACMTestISACConfig& swbISACConfig);
+
+  void SwitchingSamplingRate(int testNr, int maxSampRateChange);
+
+  rtc::scoped_ptr<AudioCodingModule> _acmA;
+  rtc::scoped_ptr<AudioCodingModule> _acmB;
+
+  rtc::scoped_ptr<Channel> _channel_A2B;
+  rtc::scoped_ptr<Channel> _channel_B2A;
+
+  PCMFile _inFileA;
+  PCMFile _inFileB;
+
+  PCMFile _outFileA;
+  PCMFile _outFileB;
+
+  uint8_t _idISAC16kHz;
+  uint8_t _idISAC32kHz;
+  CodecInst _paramISAC16kHz;
+  CodecInst _paramISAC32kHz;
+
+  std::string file_name_swb_;
+
+  ACMTestTimer _myTimer;
+  int _testMode;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_

webrtc/modules/audio_coding/test/insert_packet_with_timing.cc

@@ -0,0 +1,307 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "gflags/gflags.h"
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+// Codec.
+DEFINE_string(codec, "opus", "Codec Name");
+DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
+DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
+
+// PCM input/output.
+DEFINE_string(input, "", "Input PCM file at 16 kHz.");
+DEFINE_bool(input_stereo, false, "Input is stereo.");
+DEFINE_int32(input_fs_hz, 32000, "Input sample rate Hz.");
+DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
+DEFINE_int32(output_fs_hz, 32000, "Output sample rate Hz");
+
+// Timing files
+DEFINE_string(seq_num, "seq_num", "Sequence number file.");
+DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
+DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
+
+// Delay logging
+DEFINE_string(delay, "", "Log for delay.");
+
+// Other setups
+DEFINE_bool(verbose, false, "Verbosity.");
+DEFINE_double(loss_rate, 0, "Rate of packet loss < 1");
+
+const int32_t kAudioPlayedOut = 0x00000001;
+const int32_t kPacketPushedIn = 0x00000001 << 1;
+const int kPlayoutPeriodMs = 10;
+
+namespace webrtc {
+
+class InsertPacketWithTiming {
+ public:
+  InsertPacketWithTiming()
+      : sender_clock_(new SimulatedClock(0)),
+        receiver_clock_(new SimulatedClock(0)),
+        send_acm_(AudioCodingModule::Create(0, sender_clock_)),
+        receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
+        channel_(new Channel),
+        seq_num_fid_(fopen(FLAGS_seq_num.c_str(), "rt")),
+        send_ts_fid_(fopen(FLAGS_send_ts.c_str(), "rt")),
+        receive_ts_fid_(fopen(FLAGS_receive_ts.c_str(), "rt")),
+        pcm_out_fid_(fopen(FLAGS_output.c_str(), "wb")),
+        samples_in_1ms_(48),
+        num_10ms_in_codec_frame_(2),  // Typical 20 ms frames.
+        time_to_insert_packet_ms_(3),  // An arbitrary offset on pushing packet.
+        next_receive_ts_(0),
+        time_to_playout_audio_ms_(kPlayoutPeriodMs),
+        loss_threshold_(0),
+        playout_timing_fid_(fopen("playout_timing.txt", "wt")) {}
+
+  void SetUp() {
+    ASSERT_TRUE(sender_clock_ != NULL);
+    ASSERT_TRUE(receiver_clock_ != NULL);
+
+    ASSERT_TRUE(send_acm_.get() != NULL);
+    ASSERT_TRUE(receive_acm_.get() != NULL);
+    ASSERT_TRUE(channel_ != NULL);
+
+    ASSERT_TRUE(seq_num_fid_ != NULL);
+    ASSERT_TRUE(send_ts_fid_ != NULL);
+    ASSERT_TRUE(receive_ts_fid_ != NULL);
+
+    ASSERT_TRUE(playout_timing_fid_ != NULL);
+
+    next_receive_ts_ = ReceiveTimestamp();
+
+    CodecInst codec;
+    ASSERT_EQ(0, AudioCodingModule::Codec(FLAGS_codec.c_str(), &codec,
+                             FLAGS_codec_sample_rate_hz,
+                             FLAGS_codec_channels));
+    ASSERT_EQ(0, receive_acm_->InitializeReceiver());
+    ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
+    ASSERT_EQ(0, receive_acm_->RegisterReceiveCodec(codec));
+
+    // Set codec-dependent parameters.
+    samples_in_1ms_ = codec.plfreq / 1000;
+    num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
+
+    channel_->RegisterReceiverACM(receive_acm_.get());
+    send_acm_->RegisterTransportCallback(channel_);
+
+    if (FLAGS_input.size() == 0) {
+      std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
+                                                 "pcm");
+      pcm_in_fid_.Open(file_name, 32000, "r", true);  // auto-rewind
+      std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
+    } else {
+      pcm_in_fid_.Open(FLAGS_input, static_cast<uint16_t>(FLAGS_input_fs_hz),
+                    "r", true);  // auto-rewind
+      std::cout << "Input file " << FLAGS_input << "at " << FLAGS_input_fs_hz
+          << " Hz in " << ((FLAGS_input_stereo) ? "stereo." : "mono.")
+          << std::endl;
+      pcm_in_fid_.ReadStereo(FLAGS_input_stereo);
+    }
+
+    ASSERT_TRUE(pcm_out_fid_ != NULL);
+    std::cout << "Output file " << FLAGS_output << " at " << FLAGS_output_fs_hz
+        << " Hz." << std::endl;
+
+    // Other setups
+    if (FLAGS_loss_rate > 0)
+      loss_threshold_ = RAND_MAX * FLAGS_loss_rate;
+    else
+      loss_threshold_ = 0;
+  }
+
+  void TickOneMillisecond(uint32_t* action) {
+    // One millisecond passed.
+    time_to_insert_packet_ms_--;
+    time_to_playout_audio_ms_--;
+    sender_clock_->AdvanceTimeMilliseconds(1);
+    receiver_clock_->AdvanceTimeMilliseconds(1);
+
+    // Reset action.
+    *action = 0;
+
+    // Is it time to pull audio?
+    if (time_to_playout_audio_ms_ == 0) {
+      time_to_playout_audio_ms_ = kPlayoutPeriodMs;
+      receive_acm_->PlayoutData10Ms(static_cast<int>(FLAGS_output_fs_hz),
+                                    &frame_);
+      fwrite(frame_.data_, sizeof(frame_.data_[0]),
+             frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
+      *action |= kAudioPlayedOut;
+    }
+
+    // Is it time to push in next packet?
+    if (time_to_insert_packet_ms_ <= .5) {
+      *action |= kPacketPushedIn;
+
+      // Update time-to-insert packet.
+      uint32_t t = next_receive_ts_;
+      next_receive_ts_ = ReceiveTimestamp();
+      time_to_insert_packet_ms_ += static_cast<float>(next_receive_ts_ - t) /
+          samples_in_1ms_;
+
+      // Push in just enough audio.
+      for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
+        pcm_in_fid_.Read10MsData(frame_);
+        EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
+      }
+
+      // Set the parameters for the packet to be pushed in receiver ACM right
+      // now.
+      uint32_t ts = SendTimestamp();
+      int seq_num = SequenceNumber();
+      bool lost = false;
+      channel_->set_send_timestamp(ts);
+      channel_->set_sequence_number(seq_num);
+      if (loss_threshold_ > 0 && rand() < loss_threshold_) {
+        channel_->set_num_packets_to_drop(1);
+        lost = true;
+      }
+
+      if (FLAGS_verbose) {
+        if (!lost) {
+          std::cout << "\nInserting packet number " << seq_num
+              << " timestamp " << ts << std::endl;
+        } else {
+          std::cout << "\nLost packet number " << seq_num
+              << " timestamp " << ts << std::endl;
+        }
+      }
+    }
+  }
+
+  void TearDown() {
+    delete channel_;
+
+    fclose(seq_num_fid_);
+    fclose(send_ts_fid_);
+    fclose(receive_ts_fid_);
+    fclose(pcm_out_fid_);
+    pcm_in_fid_.Close();
+  }
+
+  ~InsertPacketWithTiming() {
+    delete sender_clock_;
+    delete receiver_clock_;
+  }
+
+  // Are there more info to simulate.
+  bool HasPackets() {
+    if (feof(seq_num_fid_) || feof(send_ts_fid_) || feof(receive_ts_fid_))
+      return false;
+    return true;
+  }
+
+  // Jitter buffer delay.
+  void Delay(int* optimal_delay, int* current_delay) {
+    NetworkStatistics statistics;
+    receive_acm_->GetNetworkStatistics(&statistics);
+    *optimal_delay = statistics.preferredBufferSize;
+    *current_delay = statistics.currentBufferSize;
+  }
+
+ private:
+  uint32_t SendTimestamp() {
+    uint32_t t;
+    EXPECT_EQ(1, fscanf(send_ts_fid_, "%u\n", &t));
+    return t;
+  }
+
+  uint32_t ReceiveTimestamp() {
+    uint32_t t;
+    EXPECT_EQ(1, fscanf(receive_ts_fid_, "%u\n", &t));
+    return t;
+  }
+
+  int SequenceNumber() {
+    int n;
+    EXPECT_EQ(1, fscanf(seq_num_fid_, "%d\n", &n));
+    return n;
+  }
+
+  // This class just creates these pointers, not deleting them. They are deleted
+  // by the associated ACM.
+  SimulatedClock* sender_clock_;
+  SimulatedClock* receiver_clock_;
+
+  rtc::scoped_ptr<AudioCodingModule> send_acm_;
+  rtc::scoped_ptr<AudioCodingModule> receive_acm_;
+  Channel* channel_;
+
+  FILE* seq_num_fid_;  // Input (text), one sequence number per line.
+  FILE* send_ts_fid_;  // Input (text), one send timestamp per line.
+  FILE* receive_ts_fid_;  // Input (text), one receive timestamp per line.
+  FILE* pcm_out_fid_;  // Output PCM16.
+
+  PCMFile pcm_in_fid_;  // Input PCM16.
+
+  int samples_in_1ms_;
+
+  // TODO(turajs): this can be computed from the send timestamp, but there is
+  // some complication to account for lost and reordered packets.
+  int num_10ms_in_codec_frame_;
+
+  float time_to_insert_packet_ms_;
+  uint32_t next_receive_ts_;
+  uint32_t time_to_playout_audio_ms_;
+
+  AudioFrame frame_;
+
+  double loss_threshold_;
+
+  // Output (text), sequence number, playout timestamp, time (ms) of playout,
+  // per line.
+  FILE* playout_timing_fid_;
+};
+
+}  // webrtc
+
+int main(int argc, char* argv[]) {
+  google::ParseCommandLineFlags(&argc, &argv, true);
+  webrtc::InsertPacketWithTiming test;
+  test.SetUp();
+
+  FILE* delay_log = NULL;
+  if (FLAGS_delay.size() > 0) {
+    delay_log = fopen(FLAGS_delay.c_str(), "wt");
+    if (delay_log == NULL) {
+      std::cout << "Cannot open the file to log delay values." << std::endl;
+      exit(1);
+    }
+  }
+
+  uint32_t action_taken;
+  int optimal_delay_ms;
+  int current_delay_ms;
+  while (test.HasPackets()) {
+    test.TickOneMillisecond(&action_taken);
+
+    if (action_taken != 0) {
+      test.Delay(&optimal_delay_ms, &current_delay_ms);
+      if (delay_log != NULL) {
+        fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
+      }
+    }
+  }
+  std::cout << std::endl;
+  test.TearDown();
+  if (delay_log != NULL)
+    fclose(delay_log);
+}

webrtc/modules/audio_coding/test/opus_test.cc

@@ -0,0 +1,380 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_coding/test/opus_test.h"
+
+#include <assert.h>
+
+#include <string>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+
+OpusTest::OpusTest()
+    : acm_receiver_(AudioCodingModule::Create(0)),
+      channel_a2b_(NULL),
+      counter_(0),
+      payload_type_(255),
+      rtp_timestamp_(0) {}
+
+OpusTest::~OpusTest() {
+  if (channel_a2b_ != NULL) {
+    delete channel_a2b_;
+    channel_a2b_ = NULL;
+  }
+  if (opus_mono_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_mono_encoder_);
+    opus_mono_encoder_ = NULL;
+  }
+  if (opus_stereo_encoder_ != NULL) {
+    WebRtcOpus_EncoderFree(opus_stereo_encoder_);
+    opus_stereo_encoder_ = NULL;
+  }
+  if (opus_mono_decoder_ != NULL) {
+    WebRtcOpus_DecoderFree(opus_mono_decoder_);
+    opus_mono_decoder_ = NULL;
+  }
+  if (opus_stereo_decoder_ != NULL) {
+    WebRtcOpus_DecoderFree(opus_stereo_decoder_);
+    opus_stereo_decoder_ = NULL;
+  }
+}
+
+void OpusTest::Perform() {
+#ifndef WEBRTC_CODEC_OPUS
+  // Opus isn't defined, exit.
+  return;
+#else
+  uint16_t frequency_hz;
+  int audio_channels;
+  int16_t test_cntr = 0;
+
+  // Open both mono and stereo test files in 32 kHz.
+  const std::string file_name_stereo =
+      webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
+  const std::string file_name_mono =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  frequency_hz = 32000;
+  in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb");
+  in_file_stereo_.ReadStereo(true);
+  in_file_mono_.Open(file_name_mono, frequency_hz, "rb");
+  in_file_mono_.ReadStereo(false);
+
+  // Create Opus encoders for mono and stereo.
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0), -1);
+  ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1), -1);
+
+  // Create Opus decoders for mono and stereo for stand-alone testing of Opus.
+  ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1);
+  ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1);
+  WebRtcOpus_DecoderInit(opus_mono_decoder_);
+  WebRtcOpus_DecoderInit(opus_stereo_decoder_);
+
+  ASSERT_TRUE(acm_receiver_.get() != NULL);
+  EXPECT_EQ(0, acm_receiver_->InitializeReceiver());
+
+  // Register Opus stereo as receiving codec.
+  CodecInst opus_codec_param;
+  int codec_id = acm_receiver_->Codec("opus", 48000, 2);
+  EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param));
+  payload_type_ = opus_codec_param.pltype;
+  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+
+  // Create and connect the channel.
+  channel_a2b_ = new TestPackStereo;
+  channel_a2b_->RegisterReceiverACM(acm_receiver_.get());
+
+  //
+  // Test Stereo.
+  //
+
+  channel_a2b_->set_codec_mode(kStereo);
+  audio_channels = 2;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 64000, 2880);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Opus stereo with packet-losses.
+  //
+
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 20 ms frame size, 1% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 1);
+
+  // Run Opus with 20 ms frame size, 5% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 5);
+
+  // Run Opus with 20 ms frame size, 10% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 10);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Mono.
+  //
+  channel_a2b_->set_codec_mode(kMono);
+  audio_channels = 1;
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Register Opus mono as receiving codec.
+  opus_codec_param.channels = 1;
+  EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param));
+
+  // Run Opus with 2.5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 120);
+
+  // Run Opus with 5 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 240);
+
+  // Run Opus with 10 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 480);
+
+  // Run Opus with 20 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 960);
+
+  // Run Opus with 40 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 1920);
+
+  // Run Opus with 60 ms frame size.
+  Run(channel_a2b_, audio_channels, 32000, 2880);
+
+  out_file_.Close();
+  out_file_standalone_.Close();
+
+  //
+  // Test Opus mono with packet-losses.
+  //
+  test_cntr++;
+  OpenOutFile(test_cntr);
+
+  // Run Opus with 20 ms frame size, 1% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 1);
+
+  // Run Opus with 20 ms frame size, 5% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 5);
+
+  // Run Opus with 20 ms frame size, 10% packet loss.
+  Run(channel_a2b_, audio_channels, 64000, 960, 10);
+
+  // Close the files.
+  in_file_stereo_.Close();
+  in_file_mono_.Close();
+  out_file_.Close();
+  out_file_standalone_.Close();
+#endif
+}
+
+void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate,
+                   int frame_length, int percent_loss) {
+  AudioFrame audio_frame;
+  int32_t out_freq_hz_b = out_file_.SamplingFrequency();
+  const int kBufferSizeSamples = 480 * 12 * 2;  // Can hold 120 ms stereo audio.
+  int16_t audio[kBufferSizeSamples];
+  int16_t out_audio[kBufferSizeSamples];
+  int16_t audio_type;
+  int written_samples = 0;
+  int read_samples = 0;
+  int decoded_samples = 0;
+  bool first_packet = true;
+  uint32_t start_time_stamp = 0;
+
+  channel->reset_payload_size();
+  counter_ = 0;
+
+  // Set encoder rate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate));
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
+  // If we are on Android, iOS and/or ARM, use a lower complexity setting as
+  // default.
+  const int kOpusComplexity5 = 5;
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
+                                        kOpusComplexity5));
+#endif
+
+  // Make sure the runtime is less than 60 seconds to pass Android test.
+  for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) {
+    bool lost_packet = false;
+
+    // Get 10 msec of audio.
+    if (channels == 1) {
+      if (in_file_mono_.EndOfFile()) {
+        break;
+      }
+      in_file_mono_.Read10MsData(audio_frame);
+    } else {
+      if (in_file_stereo_.EndOfFile()) {
+        break;
+      }
+      in_file_stereo_.Read10MsData(audio_frame);
+    }
+
+    // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
+    EXPECT_EQ(480,
+              resampler_.Resample10Msec(audio_frame.data_,
+                                        audio_frame.sample_rate_hz_,
+                                        48000,
+                                        channels,
+                                        kBufferSizeSamples - written_samples,
+                                        &audio[written_samples]));
+    written_samples += 480 * channels;
+
+    // Sometimes we need to loop over the audio vector to produce the right
+    // number of packets.
+    int loop_encode = (written_samples - read_samples) /
+        (channels * frame_length);
+
+    if (loop_encode > 0) {
+      const int kMaxBytes = 1000;  // Maximum number of bytes for one packet.
+      size_t bitstream_len_byte;
+      uint8_t bitstream[kMaxBytes];
+      for (int i = 0; i < loop_encode; i++) {
+        int bitstream_len_byte_int = WebRtcOpus_Encode(
+            (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
+            &audio[read_samples], frame_length, kMaxBytes, bitstream);
+        ASSERT_GE(bitstream_len_byte_int, 0);
+        bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
+
+        // Simulate packet loss by setting |packet_loss_| to "true" in
+        // |percent_loss| percent of the loops.
+        // TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
+        if (percent_loss > 0) {
+          if (counter_ == floor((100 / percent_loss) + 0.5)) {
+            counter_ = 0;
+            lost_packet = true;
+            channel->set_lost_packet(true);
+          } else {
+            lost_packet = false;
+            channel->set_lost_packet(false);
+          }
+          counter_++;
+        }
+
+        // Run stand-alone Opus decoder, or decode PLC.
+        if (channels == 1) {
+          if (!lost_packet) {
+            decoded_samples += WebRtcOpus_Decode(
+                opus_mono_decoder_, bitstream, bitstream_len_byte,
+                &out_audio[decoded_samples * channels], &audio_type);
+          } else {
+            decoded_samples += WebRtcOpus_DecodePlc(
+                opus_mono_decoder_, &out_audio[decoded_samples * channels], 1);
+          }
+        } else {
+          if (!lost_packet) {
+            decoded_samples += WebRtcOpus_Decode(
+                opus_stereo_decoder_, bitstream, bitstream_len_byte,
+                &out_audio[decoded_samples * channels], &audio_type);
+          } else {
+            decoded_samples += WebRtcOpus_DecodePlc(
+                opus_stereo_decoder_, &out_audio[decoded_samples * channels],
+                1);
+          }
+        }
+
+        // Send data to the channel. "channel" will handle the loss simulation.
+        channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_,
+                          bitstream, bitstream_len_byte, NULL);
+        if (first_packet) {
+          first_packet = false;
+          start_time_stamp = rtp_timestamp_;
+        }
+        rtp_timestamp_ += frame_length;
+        read_samples += frame_length * channels;
+      }
+      if (read_samples == written_samples) {
+        read_samples = 0;
+        written_samples = 0;
+      }
+    }
+
+    // Run received side of ACM.
+    ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
+
+    // Write output speech to file.
+    out_file_.Write10MsData(
+        audio_frame.data_,
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+
+    // Write stand-alone speech to file.
+    out_file_standalone_.Write10MsData(
+        out_audio, static_cast<size_t>(decoded_samples) * channels);
+
+    if (audio_frame.timestamp_ > start_time_stamp) {
+      // Number of channels should be the same for both stand-alone and
+      // ACM-decoding.
+      EXPECT_EQ(audio_frame.num_channels_, channels);
+    }
+
+    decoded_samples = 0;
+  }
+
+  if (in_file_mono_.EndOfFile()) {
+    in_file_mono_.Rewind();
+  }
+  if (in_file_stereo_.EndOfFile()) {
+    in_file_stereo_.Rewind();
+  }
+  // Reset in case we ended with a lost packet.
+  channel->set_lost_packet(false);
+}
+
+void OpusTest::OpenOutFile(int test_number) {
+  std::string file_name;
+  std::stringstream file_stream;
+  file_stream << webrtc::test::OutputPath() << "opustest_out_"
+      << test_number << ".pcm";
+  file_name = file_stream.str();
+  out_file_.Open(file_name, 48000, "wb");
+  file_stream.str("");
+  file_name = file_stream.str();
+  file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
+      << test_number << ".pcm";
+  file_name = file_stream.str();
+  out_file_standalone_.Open(file_name, 48000, "wb");
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/opus_test.h

@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+
+#include <math.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "webrtc/modules/audio_coding/test/TestStereo.h"
+
+namespace webrtc {
+
+class OpusTest : public ACMTest {
+ public:
+  OpusTest();
+  ~OpusTest();
+
+  void Perform();
+
+ private:
+  void Run(TestPackStereo* channel, int channels, int bitrate, int frame_length,
+           int percent_loss = 0);
+
+  void OpenOutFile(int test_number);
+
+  rtc::scoped_ptr<AudioCodingModule> acm_receiver_;
+  TestPackStereo* channel_a2b_;
+  PCMFile in_file_stereo_;
+  PCMFile in_file_mono_;
+  PCMFile out_file_;
+  PCMFile out_file_standalone_;
+  int counter_;
+  uint8_t payload_type_;
+  int rtp_timestamp_;
+  acm2::ACMResampler resampler_;
+  WebRtcOpusEncInst* opus_mono_encoder_;
+  WebRtcOpusEncInst* opus_stereo_encoder_;
+  WebRtcOpusDecInst* opus_mono_decoder_;
+  WebRtcOpusDecInst* opus_stereo_decoder_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_

webrtc/modules/audio_coding/test/target_delay_unittest.cc

@@ -0,0 +1,223 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/test/utility.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
+
+namespace webrtc {
+
+class TargetDelayTest : public ::testing::Test {
+ protected:
+  TargetDelayTest() : acm_(AudioCodingModule::Create(0)) {}
+
+  ~TargetDelayTest() {}
+
+  void SetUp() {
+    EXPECT_TRUE(acm_.get() != NULL);
+
+    CodecInst codec;
+    ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1));
+    ASSERT_EQ(0, acm_->InitializeReceiver());
+    ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec));
+
+    rtp_info_.header.payloadType = codec.pltype;
+    rtp_info_.header.timestamp = 0;
+    rtp_info_.header.ssrc = 0x12345678;
+    rtp_info_.header.markerBit = false;
+    rtp_info_.header.sequenceNumber = 0;
+    rtp_info_.type.Audio.channel = 1;
+    rtp_info_.type.Audio.isCNG = false;
+    rtp_info_.frameType = kAudioFrameSpeech;
+
+    int16_t audio[kFrameSizeSamples];
+    const int kRange = 0x7FF;  // 2047, easy for masking.
+    for (size_t n = 0; n < kFrameSizeSamples; ++n)
+      audio[n] = (rand() & kRange) - kRange / 2;
+    WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
+  }
+
+  void OutOfRangeInput() {
+    EXPECT_EQ(-1, SetMinimumDelay(-1));
+    EXPECT_EQ(-1, SetMinimumDelay(10001));
+  }
+
+  void NoTargetDelayBufferSizeChanges() {
+    for (int n = 0; n < 30; ++n)  // Run enough iterations.
+      Run(true);
+    int clean_optimal_delay = GetCurrentOptimalDelayMs();
+    Run(false);  // Run with jitter.
+    int jittery_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_GT(jittery_optimal_delay, clean_optimal_delay);
+    int required_delay = RequiredDelay();
+    EXPECT_GT(required_delay, 0);
+    EXPECT_NEAR(required_delay, jittery_optimal_delay, 1);
+  }
+
+  void WithTargetDelayBufferNotChanging() {
+    // A target delay that is one packet larger than jitter.
+    const int kTargetDelayMs = (kInterarrivalJitterPacket + 1) *
+        kNum10msPerFrame * 10;
+    ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
+    for (int n = 0; n < 30; ++n)  // Run enough iterations to fill the buffer.
+      Run(true);
+    int clean_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetDelayMs, clean_optimal_delay);
+    Run(false);  // Run with jitter.
+    int jittery_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(jittery_optimal_delay, clean_optimal_delay);
+  }
+
+  void RequiredDelayAtCorrectRange() {
+    for (int n = 0; n < 30; ++n)  // Run clean and store delay.
+      Run(true);
+    int clean_optimal_delay = GetCurrentOptimalDelayMs();
+
+    // A relatively large delay.
+    const int kTargetDelayMs = (kInterarrivalJitterPacket + 10) *
+        kNum10msPerFrame * 10;
+    ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
+    for (int n = 0; n < 300; ++n)  // Run enough iterations to fill the buffer.
+      Run(true);
+    Run(false);  // Run with jitter.
+
+    int jittery_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetDelayMs, jittery_optimal_delay);
+
+    int required_delay = RequiredDelay();
+
+    // Checking |required_delay| is in correct range.
+    EXPECT_GT(required_delay, 0);
+    EXPECT_GT(jittery_optimal_delay, required_delay);
+    EXPECT_GT(required_delay, clean_optimal_delay);
+
+    // A tighter check for the value of |required_delay|.
+    // The jitter forces a delay of
+    // |kInterarrivalJitterPacket * kNum10msPerFrame * 10| milliseconds. So we
+    // expect |required_delay| be close to that.
+    EXPECT_NEAR(kInterarrivalJitterPacket * kNum10msPerFrame * 10,
+                required_delay, 1);
+  }
+
+  void TargetDelayBufferMinMax() {
+    const int kTargetMinDelayMs = kNum10msPerFrame * 10;
+    ASSERT_EQ(0, SetMinimumDelay(kTargetMinDelayMs));
+    for (int m = 0; m < 30; ++m)  // Run enough iterations to fill the buffer.
+      Run(true);
+    int clean_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetMinDelayMs, clean_optimal_delay);
+
+    const int kTargetMaxDelayMs = 2 * (kNum10msPerFrame * 10);
+    ASSERT_EQ(0, SetMaximumDelay(kTargetMaxDelayMs));
+    for (int n = 0; n < 30; ++n)  // Run enough iterations to fill the buffer.
+      Run(false);
+
+    int capped_optimal_delay = GetCurrentOptimalDelayMs();
+    EXPECT_EQ(kTargetMaxDelayMs, capped_optimal_delay);
+  }
+
+ private:
+  static const int kSampleRateHz = 16000;
+  static const int kNum10msPerFrame = 2;
+  static const size_t kFrameSizeSamples = 320;  // 20 ms @ 16 kHz.
+  // payload-len = frame-samples * 2 bytes/sample.
+  static const int kPayloadLenBytes = 320 * 2;
+  // Inter-arrival time in number of packets in a jittery channel. One is no
+  // jitter.
+  static const int kInterarrivalJitterPacket = 2;
+
+  void Push() {
+    rtp_info_.header.timestamp += kFrameSizeSamples;
+    rtp_info_.header.sequenceNumber++;
+    ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
+                                      rtp_info_));
+  }
+
+  // Pull audio equivalent to the amount of audio in one RTP packet.
+  void Pull() {
+    AudioFrame frame;
+    for (int k = 0; k < kNum10msPerFrame; ++k) {  // Pull one frame.
+      ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
+      // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
+      ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
+      ASSERT_EQ(1, frame.num_channels_);
+      ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
+    }
+  }
+
+  void Run(bool clean) {
+    for (int n = 0; n < 10; ++n) {
+      for (int m = 0; m < 5; ++m) {
+        Push();
+        Pull();
+      }
+
+      if (!clean) {
+        for (int m = 0; m < 10; ++m) {  // Long enough to trigger delay change.
+          Push();
+          for (int n = 0; n < kInterarrivalJitterPacket; ++n)
+            Pull();
+        }
+      }
+    }
+  }
+
+  int SetMinimumDelay(int delay_ms) {
+    return acm_->SetMinimumPlayoutDelay(delay_ms);
+  }
+
+  int SetMaximumDelay(int delay_ms) {
+    return acm_->SetMaximumPlayoutDelay(delay_ms);
+  }
+
+  int GetCurrentOptimalDelayMs() {
+    NetworkStatistics stats;
+    acm_->GetNetworkStatistics(&stats);
+    return stats.preferredBufferSize;
+  }
+
+  int RequiredDelay() {
+    return acm_->LeastRequiredDelayMs();
+  }
+
+  rtc::scoped_ptr<AudioCodingModule> acm_;
+  WebRtcRTPHeader rtp_info_;
+  uint8_t payload_[kPayloadLenBytes];
+};
+
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(OutOfRangeInput)) {
+  OutOfRangeInput();
+}
+
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(NoTargetDelayBufferSizeChanges)) {
+  NoTargetDelayBufferSizeChanges();
+}
+
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(WithTargetDelayBufferNotChanging)) {
+  WithTargetDelayBufferNotChanging();
+}
+
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(RequiredDelayAtCorrectRange)) {
+  RequiredDelayAtCorrectRange();
+}
+
+TEST_F(TargetDelayTest, DISABLED_ON_ANDROID(TargetDelayBufferMinMax)) {
+  TargetDelayBufferMinMax();
+}
+
+}  // namespace webrtc
+

webrtc/modules/audio_coding/test/utility.cc

@@ -0,0 +1,303 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "utility.h"
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/common.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+
+#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
+
+namespace webrtc {
+
+ACMTestTimer::ACMTestTimer()
+    : _msec(0),
+      _sec(0),
+      _min(0),
+      _hour(0) {
+  return;
+}
+
+ACMTestTimer::~ACMTestTimer() {
+  return;
+}
+
+void ACMTestTimer::Reset() {
+  _msec = 0;
+  _sec = 0;
+  _min = 0;
+  _hour = 0;
+  return;
+}
+void ACMTestTimer::Tick10ms() {
+  _msec += 10;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick1ms() {
+  _msec++;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick100ms() {
+  _msec += 100;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::Tick1sec() {
+  _sec++;
+  Adjust();
+  return;
+}
+
+void ACMTestTimer::CurrentTimeHMS(char* currTime) {
+  sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min,
+          (double) _sec + (double) _msec / 1000.);
+  return;
+}
+
+void ACMTestTimer::CurrentTime(unsigned long& h, unsigned char& m,
+                               unsigned char& s, unsigned short& ms) {
+  h = _hour;
+  m = _min;
+  s = _sec;
+  ms = _msec;
+  return;
+}
+
+void ACMTestTimer::Adjust() {
+  unsigned int n;
+  if (_msec >= 1000) {
+    n = _msec / 1000;
+    _msec -= (1000 * n);
+    _sec += n;
+  }
+  if (_sec >= 60) {
+    n = _sec / 60;
+    _sec -= (n * 60);
+    _min += n;
+  }
+  if (_min >= 60) {
+    n = _min / 60;
+    _min -= (n * 60);
+    _hour += n;
+  }
+}
+
+int16_t ChooseCodec(CodecInst& codecInst) {
+
+  PrintCodecs();
+  //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
+  uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+  int8_t codecID;
+  bool outOfRange = false;
+  char myStr[15] = "";
+  do {
+    printf("\nChoose a codec [0]: ");
+    EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
+    codecID = atoi(myStr);
+    if ((codecID < 0) || (codecID >= noCodec)) {
+      printf("\nOut of range.\n");
+      outOfRange = true;
+    }
+  } while (outOfRange);
+
+  CHECK_ERROR(AudioCodingModule::Codec((uint8_t )codecID, &codecInst));
+  return 0;
+}
+
+void PrintCodecs() {
+  uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
+
+  CodecInst codecInst;
+  printf("No  Name                [Hz]    [bps]\n");
+  for (uint8_t codecCntr = 0; codecCntr < noCodec; codecCntr++) {
+    AudioCodingModule::Codec(codecCntr, &codecInst);
+    printf("%2d- %-18s %5d   %6d\n", codecCntr, codecInst.plname,
+           codecInst.plfreq, codecInst.rate);
+  }
+
+}
+
+CircularBuffer::CircularBuffer(uint32_t len)
+    : _buff(NULL),
+      _idx(0),
+      _buffIsFull(false),
+      _calcAvg(false),
+      _calcVar(false),
+      _sum(0),
+      _sumSqr(0) {
+  _buff = new double[len];
+  if (_buff == NULL) {
+    _buffLen = 0;
+  } else {
+    for (uint32_t n = 0; n < len; n++) {
+      _buff[n] = 0;
+    }
+    _buffLen = len;
+  }
+}
+
+CircularBuffer::~CircularBuffer() {
+  if (_buff != NULL) {
+    delete[] _buff;
+    _buff = NULL;
+  }
+}
+
+void CircularBuffer::Update(const double newVal) {
+  assert(_buffLen > 0);
+
+  // store the value that is going to be overwritten
+  double oldVal = _buff[_idx];
+  // record the new value
+  _buff[_idx] = newVal;
+  // increment the index, to point to where we would
+  // write next
+  _idx++;
+  // it is a circular buffer, if we are at the end
+  // we have to cycle to the beginning
+  if (_idx >= _buffLen) {
+    // flag that the buffer is filled up.
+    _buffIsFull = true;
+    _idx = 0;
+  }
+
+  // Update
+
+  if (_calcAvg) {
+    // for the average we have to update
+    // the sum
+    _sum += (newVal - oldVal);
+  }
+
+  if (_calcVar) {
+    // to calculate variance we have to update
+    // the sum of squares
+    _sumSqr += (double) (newVal - oldVal) * (double) (newVal + oldVal);
+  }
+}
+
+void CircularBuffer::SetArithMean(bool enable) {
+  assert(_buffLen > 0);
+
+  if (enable && !_calcAvg) {
+    uint32_t lim;
+    if (_buffIsFull) {
+      lim = _buffLen;
+    } else {
+      lim = _idx;
+    }
+    _sum = 0;
+    for (uint32_t n = 0; n < lim; n++) {
+      _sum += _buff[n];
+    }
+  }
+  _calcAvg = enable;
+}
+
+void CircularBuffer::SetVariance(bool enable) {
+  assert(_buffLen > 0);
+
+  if (enable && !_calcVar) {
+    uint32_t lim;
+    if (_buffIsFull) {
+      lim = _buffLen;
+    } else {
+      lim = _idx;
+    }
+    _sumSqr = 0;
+    for (uint32_t n = 0; n < lim; n++) {
+      _sumSqr += _buff[n] * _buff[n];
+    }
+  }
+  _calcAvg = enable;
+}
+
+int16_t CircularBuffer::ArithMean(double& mean) {
+  assert(_buffLen > 0);
+
+  if (_buffIsFull) {
+
+    mean = _sum / (double) _buffLen;
+    return 0;
+  } else {
+    if (_idx > 0) {
+      mean = _sum / (double) _idx;
+      return 0;
+    } else {
+      return -1;
+    }
+
+  }
+}
+
+int16_t CircularBuffer::Variance(double& var) {
+  assert(_buffLen > 0);
+
+  if (_buffIsFull) {
+    var = _sumSqr / (double) _buffLen;
+    return 0;
+  } else {
+    if (_idx > 0) {
+      var = _sumSqr / (double) _idx;
+      return 0;
+    } else {
+      return -1;
+    }
+  }
+}
+
+bool FixedPayloadTypeCodec(const char* payloadName) {
+  char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = { "PCMU",
+      "PCMA", "GSM", "G723", "DVI4", "LPC", "PCMA", "G722", "QCELP", "CN",
+      "MPA", "G728", "G729" };
+
+  for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
+    if (!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n])) {
+      return true;
+    }
+  }
+  return false;
+}
+
+void VADCallback::Reset() {
+  memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+VADCallback::VADCallback() {
+  memset(_numFrameTypes, 0, sizeof(_numFrameTypes));
+}
+
+void VADCallback::PrintFrameTypes() {
+  printf("kEmptyFrame......... %d\n", _numFrameTypes[kEmptyFrame]);
+  printf("kAudioFrameSpeech... %d\n", _numFrameTypes[kAudioFrameSpeech]);
+  printf("kAudioFrameCN....... %d\n", _numFrameTypes[kAudioFrameCN]);
+  printf("kVideoFrameKey...... %d\n", _numFrameTypes[kVideoFrameKey]);
+  printf("kVideoFrameDelta.... %d\n", _numFrameTypes[kVideoFrameDelta]);
+}
+
+int32_t VADCallback::InFrameType(FrameType frame_type) {
+  _numFrameTypes[frame_type]++;
+  return 0;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_coding/test/utility.h

@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+
+namespace webrtc {
+
+//-----------------------------
+#define CHECK_ERROR(f)                                                         \
+  do {                                                                         \
+    EXPECT_GE(f, 0) << "Error Calling API";                                    \
+  } while(0)
+
+//-----------------------------
+#define CHECK_PROTECTED(f)                                                     \
+  do {                                                                         \
+    if (f >= 0) {                                                              \
+      ADD_FAILURE() << "Error Calling API";                                    \
+    } else {                                                                   \
+      printf("An expected error is caught.\n");                                \
+    }                                                                          \
+  } while(0)
+
+//----------------------------
+#define CHECK_ERROR_MT(f)                                                      \
+  do {                                                                         \
+    if (f < 0) {                                                               \
+      fprintf(stderr, "Error Calling API in file %s at line %d \n",            \
+              __FILE__, __LINE__);                                             \
+    }                                                                          \
+  } while(0)
+
+//----------------------------
+#define CHECK_PROTECTED_MT(f)                                                  \
+  do {                                                                         \
+    if (f >= 0) {                                                              \
+      fprintf(stderr, "Error Calling API in file %s at line %d \n",            \
+              __FILE__, __LINE__);                                             \
+    } else {                                                                   \
+      printf("An expected error is caught.\n");                                \
+    }                                                                          \
+  } while(0)
+
+#define DELETE_POINTER(p)                                                      \
+  do {                                                                         \
+    if (p != NULL) {                                                           \
+      delete p;                                                                \
+      p = NULL;                                                                \
+    }                                                                          \
+  } while(0)
+
+class ACMTestTimer {
+ public:
+  ACMTestTimer();
+  ~ACMTestTimer();
+
+  void Reset();
+  void Tick10ms();
+  void Tick1ms();
+  void Tick100ms();
+  void Tick1sec();
+  void CurrentTimeHMS(char* currTime);
+  void CurrentTime(unsigned long& h, unsigned char& m, unsigned char& s,
+                   unsigned short& ms);
+
+ private:
+  void Adjust();
+
+  unsigned short _msec;
+  unsigned char _sec;
+  unsigned char _min;
+  unsigned long _hour;
+};
+
+class CircularBuffer {
+ public:
+  CircularBuffer(uint32_t len);
+  ~CircularBuffer();
+
+  void SetArithMean(bool enable);
+  void SetVariance(bool enable);
+
+  void Update(const double newVal);
+  void IsBufferFull();
+
+  int16_t Variance(double& var);
+  int16_t ArithMean(double& mean);
+
+ protected:
+  double* _buff;
+  uint32_t _idx;
+  uint32_t _buffLen;
+
+  bool _buffIsFull;
+  bool _calcAvg;
+  bool _calcVar;
+  double _sum;
+  double _sumSqr;
+};
+
+int16_t ChooseCodec(CodecInst& codecInst);
+
+void PrintCodecs();
+
+bool FixedPayloadTypeCodec(const char* payloadName);
+
+class VADCallback : public ACMVADCallback {
+ public:
+  VADCallback();
+  ~VADCallback() {
+  }
+
+  int32_t InFrameType(FrameType frame_type);
+
+  void PrintFrameTypes();
+  void Reset();
+
+ private:
+  uint32_t _numFrameTypes[5];
+};
+
+void UseLegacyAcm(webrtc::Config* config);
+
+void UseNewAcm(webrtc::Config* config);
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_